yep its registered but never invoked when calling the GK from the CME SIP phone. It is invoked when a call comes into the CME phone from CUCM and I can see it in "sh sccp conn". I am using a 7975 phone as the SIP phone on CME.
Thanks, DA On Tue, Oct 5, 2010 at 6:49 PM, Stutz, Bernhard <st...@pandacom.de> wrote: > Are you sure that your transcoder on cme is been registered? > "show sdspfarm units" will show you that. > > as far as i know you don't need any special command on the voice > register global to have the dspfarm resources beeing invoked. > > hth, > Bernhard > > ------------------------------ > *Von:* David A [mailto:david.a...@gmail.com] > *Gesendet:* Di 05.10.2010 21:08 > > *An:* Stutz, Bernhard > *Cc:* CCIE Voice GMAIL; osl osl > *Betreff:* Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls > > Yes Bernhard, When I change the codec on the voice register pool to g729 > (default) it works fine. But I have a transcoder configured on the CME on > telephony service which should be invoked if needed. The voice-class codec > is already on the outgoing dialpeer towards gk but still it does not invoke > a transcoder. I am not sure but do I need any special command on the voice > register global to invoke the transcoder? > > Thanks, > DA > > On Tue, Oct 5, 2010 at 1:40 PM, Stutz, Bernhard <st...@pandacom.de> wrote: > >> hm sounds like an codec issue... >> you have g711ulaw hardcoded at your cme sip phone. try to use there the >> voice class codec aswell or if this doesn't help add a transcoder at cme >> site aswell. >> or try to use hardcoded g729 on the sip phone pool >> don't forget to do always create prof and reset at voice register global >> after a change >> >> hth, >> Bernhard >> >> ------------------------------ >> *Von:* David A [mailto:david.a...@gmail.com] >> *Gesendet:* Di 05.10.2010 18:16 >> >> *An:* Stutz, Bernhard >> *Cc:* CCIE Voice GMAIL; osl osl >> *Betreff:* Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls >> >> Hi Bernhard, >> >> I added an MTP and a Transcoder on HQ. Added these to all mrgls. Checked >> the MTP required box and inbound faststart. When I answer the call it just >> disconnects. >> I still see the call on the GK with 16kbps coming in. >> >> Thanks, >> DA >> >> >> >> >> On Tue, Oct 5, 2010 at 11:54 AM, Stutz, Bernhard <st...@pandacom.de> >> wrote: >> >>> Hi David, >>> >>> Do you have MTP on the gk trunk enabled and inbound faststart? >>> You need to use the IOS MTP as the CUCM MTP doesn't support G.729 >>> >>> hth, >>> Bernhard >>> >>> ------------------------------ >>> *Von:* David A [mailto:david.a...@gmail.com] >>> *Gesendet:* Di 05.10.2010 17:42 >>> *An:* Stutz, Bernhard >>> *Cc:* CCIE Voice GMAIL; osl osl >>> >>> *Betreff:* Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls >>> >>> Hi Bernhard, >>> >>> The outboud call from the CME SIP phone is using the dial-peer >>> >>> dial-peer voice 15 voip >>> destination-pattern [15]...$ >>> voice-class codec 1 >>> session target ras >>> incoming called-number . >>> tech-prefix 1# >>> dtmf-relay h245-alphanumeric rtp-nte >>> When I place a call I get this >>> >>> 3845-CME-SiteC#show call active voice compact >>> <callID> A/O FAX T<sec> Codec type Peer Address IP >>> R<ip>:<udp> >>> Total call-legs: 2 >>> 290 ANS T4 g711ulaw VOIP P3002 >>> 10.10.202.54:25500 >>> 291 ORG T4 g729r8 pre- VOIP P1#1001 >>> 0.0.0.0:0 >>> >>> The other end is the GK and call ends on the SiteB phone. I dont think I >>> need a dial-peer on the GK to route to CUCM as it is done through the GK >>> trunk. >>> I can answer the call and see it on GK >>> >>> 2811-HQ-GW#sh gatekeeper call >>> Total number of active calls = 1. >>> GATEKEEPER CALL INFO >>> ==================== >>> LocalCallID Age(secs) BW >>> 51-29194 7 16(Kbps) >>> Endpt(s): Alias E.164Addr >>> src EP: SiteC-GW 3002 >>> 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#1001 >>> CallSignalAddr Port RASSignalAddr Port >>> 10.137.151.26 1720 10.137.151.26 32796 >>> But after answering there is no audio and call drops after a few seconds. >>> >>> >>> Thanks, >>> DA >>> >>> >>> >>> >>> On Tue, Oct 5, 2010 at 4:14 AM, Stutz, Bernhard <st...@pandacom.de> >>> wrote: >>> >>>> Hi, >>>> >>>> >>>> >>>> You are probably hitting the 0 dial-peer. Make sure you have a inbound >>>> dial-peer on the other end. >>>> >>>> Have a look which dial-peers you are using: >>>> >>>> >>>> >>>> sh call active voice compact >>>> >>>> or >>>> >>>> sh call active voice brief >>>> >>>> >>>> >>>> hth, >>>> >>>> Bernhard >>>> >>>> >>>> >>>> *Von:* ccie_voice-boun...@onlinestudylist.com [mailto: >>>> ccie_voice-boun...@onlinestudylist.com] *Im Auftrag von *CCIE Voice >>>> GMAIL >>>> *Gesendet:* Montag, 4. Oktober 2010 23:24 >>>> *An:* 'osl osl' >>>> *Betreff:* Re: [OSL | CCIE_Voice] Vol2 Lab2 5.1 - 4digit GK calls >>>> >>>> >>>> >>>> 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 >>>> >>>> _______________________________________________ >>>> For more information regarding industry leading CCIE Lab training, >>>> please visit www.ipexpert.com >>>> >>>> >>> >> >
_______________________________________________ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
