Re: [OSL | CCIE_Voice] QOS LFI and BACD files

Hello AJ, regarding BACD, i would you the Bulid-In Scripts inside IOS. /Robert Von: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] Im Auftrag von AJ BG Gesendet: Tuesday, February 07, 2012 7:03 AM An: ccie_voice@onlinestudylist.com Betreff: [OSL |

Re: [OSL | CCIE_Voice] IOS Version

Hi Gurpreet, this List is for the old Version 2 of the LAB. I did my preparation with 12.4(20)T or 12.4(22)T something. Unfortunately I do not find the actual List, at the Moment. There is officially stated that they use 12.4T and not 12.4 MainLine. /Robert Von: ccie_voice-boun...@onlin

Re: [OSL | CCIE_Voice] [EMEA Cluster & Gateway Dial Plan Assistance Munich Germany]

Hi, yes, we are using a "funny" dialplan here in Germany, but is not so bad. Here are some Basics: Typical PSTN Access Code: "0" LongDistance Access Code: "0" International Access Code: "00" Ok, lets say you are located in Hamburg and you want to call somebody else in Hamburg (Area Code 40), yo

Re: [OSL | CCIE_Voice] lab - (6) more MGCP TS

Hi, what do you mean by " what we can do for one way audio explanation on debugs" ? 1) Are you looking for a strategy for debugging one way audio for MGCP Gateways? 2) Or, are you looking for how to find the appropriate MGCP Message for one way audio? Is the One Way Audio an error, which we hav

Re: [OSL | CCIE_Voice] lab - 6

? /Robert Von: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] Im Auftrag von Mohammed Al Baqari Gesendet: Montag, 21. November 2011 12:14 An: 'Robert Schuknecht'; 'Abel ...'; 'datucha123 datucha123' Cc: ccie_voice@onlines

Re: [OSL | CCIE_Voice] lab - 6

Hi, SIP early offer is similar to H323 Fast-Start. With SIP EO the INVITE includes the SDP with the possible Media Options, Codecs for example. Without an NTP/Transcoder CUCM is not able to do EO on SIP Trunks. But, regarding the DHCP Part, there I am totally lost. I know how to configure D

Re: [OSL | CCIE_Voice] no dspfarm or dsparm under voice-card

Hi Bruno, Voice card 0 No dspfarm This means that you will not share your On Board DSPs with other Modules (eg. NM-HDV-2). Voice card 0 Dspfarm This means you share your On-Board DSPs with other Modules Voie card 0 Dsp ser dspfarm This means that your On-Board DSPs cou

Re: [OSL | CCIE_Voice] Voice Class Codec Question.

Hi, your configuration is ok. You will define the codec, which is actually used, inside CUCM with Region Settings. And Settings in the Enterprise and Service Parameters, as well. /Robert Von: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] Im Auftr

Re: [OSL | CCIE_Voice] RSVP

Hi, i would say double check your region settings. It seems that cucm wants to use G711 codec. Inside your dspfarm, you are missing the codec passthrough command. Is this by accident? As far as I know the RSVP locations, inside Callmanager, should have unlimited BW assigned. But, first of all

Re: [OSL | CCIE_Voice] Voice Gateway and Signaling

I think you are correct. /Robert Von: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] Im Auftrag von Ccie Voice Gesendet: Freitag, 21. Oktober 2011 12:01 An: CCIE Study Betreff: [OSL | CCIE_Voice] Voice Gateway and Signaling Hi everybody, If a

Re: [OSL | CCIE_Voice] RTMT for Unity Connection is not working

...@pcmallservices.com] Gesendet: Donnerstag, 13. Oktober 2011 23:31 An: Samia Berkane; Robert Schuknecht Cc: CCIE Study Betreff: RE: [OSL | CCIE_Voice] RTMT for Unity Connection is not working The RTMT for Cisco Unity Connection does have a problem. If you are using the plain vanilla 7.0.1 then it

Re: [OSL | CCIE_Voice] RTMT for Unity Connection is not working

. Oktober 2011 19:07 An: Robert Schuknecht Cc: CCIE Study Betreff: Re: [OSL | CCIE_Voice] RTMT for Unity Connection is not working Hi Robert, For what I remember, RTMT Tool for CUC is the same as for CUCM. When you lounch the toot, enter your Unity Connection ip address. And admin account (not

[OSL | CCIE_Voice] When will the 5-LAB Handbook be available?

Hi List, hi Vik, Last Friday I ordered the new 5-LAB Handbook, but it is still not in my Account. Is this Handbook still not released yet? If so, when will it be released? Kind Regards, Robert ___ For more information regarding industry leadi

[OSL | CCIE_Voice] RTMT for Unity Connection is not working

Hi List, I just downloaded the RTMT Plugin from my Unity Connection Server (HomeLAB) and installed it on my PC. When I start it I can not see a section for Unity Connection, in the Menu. Does anybody know how to fix this? Another thing that confuses is, that the Title Bar of the RTMT states: ..

Re: [OSL | CCIE_Voice] Phoneview and CME

Hi, try with: telephony-service url authentication http://10.10.110.3/CCMCIP/authenticate.asp admin 142536 <- same as xml user /Robert Von: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] Im Au

Re: [OSL | CCIE_Voice] Voiceview Express: Phone authentication is not working - How to debug this? -

ffice?" The 79X1 Phones are all running the same SW Vers (SCCP41.8-3-3S). /Robert -Ursprüngliche Nachricht- Von: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] Im Auftrag von Ashraf Ayyash Gesendet: Samstag, 8. Oktober 2011 00:08 An: Robert Sch

Re: [OSL | CCIE_Voice] WG: Voiceview Express: It`s getting weird

Hi Samia, yes, the 7941 and the 7961 are using the same version. It seems to be that the Phones wihich are not local to CME are functioning, besides the 7940… /Robert Von: Samia Berkane [mailto:sberkane.lear...@gmail.com] Gesendet: Freitag, 7. Oktober 2011 20:00 An: Robert Schuknecht

[OSL | CCIE_Voice] WG: Voiceview Express: It`s getting weird

my Home Office, VoiceView is working, also. So the Question is: Why is VoiceView not working when I use my 7961 Phones at my Home Office? /Robert Von: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] Im Auftrag von Robert Schuknecht Gesendet: Freitag

Re: [OSL | CCIE_Voice] Voiceview Express: Phone authentication is not working - How to debug this? -

| CCIE_Voice] Voiceview Express: Phone authentication is not working - How to debug this? - To resolve this issue, Assign phones to JTAPI user configured for CUE. On 10/6/2011 2:07 PM, Robert Schuknecht wrote: Hi, Today I tried to get Voiceview Express working, without luck. As far as I can see

[OSL | CCIE_Voice] Voiceview Express: Phone authentication is not working - How to debug this? -

Hi, Today I tried to get Voiceview Express working, without luck. As far as I can see in sniffer traces the Phone authentication is not working. And now I need some help to find my error. I already read the archive and the CUE Admin Guide, but I am not able to find the right solution. Any help

Re: [OSL | CCIE_Voice] WB1 LAB 9A BACD: Call Redirect after 90 Sec. not working -SOLVED-

Mohammed, do you know any good document which describes the timers and their relationship? /Robert Von: Mohammed Al baqari [mailto:baqari.voic...@gmail.com] Gesendet: Mittwoch, 5. Oktober 2011 13:30 An: Robert Schuknecht Cc: Geoghegan, Stuart; ccie_voice@onlinestudylist.com Betreff

Re: [OSL | CCIE_Voice] WB1 LAB 9A BACD: Call Redirect after 90 Sec. not working -SOLVED-

nswer, the call will be redirected after approx.. 3 Min. /Robert Von: Geoghegan, Stuart [mailto:stuart.geoghe...@ngbailey.co.uk] Gesendet: Dienstag, 4. Oktober 2011 17:06 An: Robert Schuknecht; ccie_voice@onlinestudylist.com Betreff: RE: [OSL | CCIE_Voice] WB1 LAB 9A BACD: Call Redirect af

Re: [OSL | CCIE_Voice] WB1 LAB 9A BACD: Call Redirect after 90 Sec. not working

mailto:stuart.geoghe...@ngbailey.co.uk] Gesendet: Dienstag, 4. Oktober 2011 16:12 An: Robert Schuknecht; ccie_voice@onlinestudylist.com Betreff: RE: [OSL | CCIE_Voice] WB1 LAB 9A BACD: Call Redirect after 90 Sec. not working Hi Robert I think that your param max-time-call-retr

[OSL | CCIE_Voice] WB1 LAB 9A BACD: Call Redirect after 90 Sec. not working

Hi, In WB 1 Question 9.2 asks us that in incoming call, into the AA 3500, should be redirected to 3001 after 90 Sec. of Ringing. I configured the BACD according to the PG but the call will be redirected after approx.. 3 Min and not after 90 Sec. Here is the relevant configuration: a

Re: [OSL | CCIE_Voice] ISDN Configuration

Hi, in my understanding they mean that you have to use clock source line under the controller (default), network-clock-participate and network-clock-select in global config, for the specific controller. Additional they want you to use isdn protocol-emulate user under the ISDN D-Channel (Serial

Re: [OSL | CCIE_Voice] Forwarded number display on branch phone

Hi, how are the numbers displayed when they hit the Branch GW? Do have have an deb isdn 931 from the Branch? /Robert Von: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] Im Auftrag von Rahul Kapor Gesendet: Freitag, 26. November 2010 18:33 An: ccie

[OSL | CCIE_Voice] Antw: v2 - IPCC aware of MWI

Interesting one... maybe send the mwi to IPCC,also. Set a variable and work on that...? I have to play with it. /Robert >>> Sergio Polizer schrieb am Dienstag, 26. Mai 2009 um >>> 21:04 in Nachricht 89d2637a1823c733028868c479af3899: > Hi, > > Someone can see a way to IPCC be aware of a MWI s

[OSL | CCIE_Voice] Antw: CCME router issue

Sanjib, yes the IOS should support both of them SRST and CME. For a real world implementaion i would ask Cisco to change the SRST License into a CME License. It should not cost anything. /Robert >>> sanjib haldar schrieb am Dienstag, 26. Mai >>> 2009 um 08:03 in Nachricht 060f36af9d2d6bb0b516

[OSL | CCIE_Voice] Antw: LAB Access: Problem with VPN/Internet - SOLVED -

It seems that there was a problem between Providers in DE and US. It now functioning again. /Robert >>> Robert Schuknecht schrieb am Freitag, 22. Mai 2009 um >>> 18:53 in Nachricht a75b6987393c831d111bd86d4723ed64: > Hi List, > > i can not access the Proctorla

[OSL | CCIE_Voice] LAB Access: Problem with VPN/Internet

Hi List, i can not access the Proctorlabs Web-Site. Also i can not access the vRacks via VPN. Is anyone else encountering this? /Robert

[OSL | CCIE_Voice] Antw: RE: QOS again: is cRTP supported with MLPPP

ginal Message- > From: ccie_voice-boun...@onlinestudylist.com > [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Robert > Schuknecht > Sent: Thursday, May 21, 2009 6:45 AM > To: OSL CCIE Voice Lab Exam > Subject: [OSL | CCIE_Voice] QOS again: is c

[OSL | CCIE_Voice] Antw: Re: what is the difference between H225GKcontrolled trunk and GK controlled Inter-cluster trunk ?

Hi, regarding the CCM 4.1 SRND on Page 8-27, an GK controlled ICT should be used to connect an CCM with an CME. "• Configure a gatekeeper-controlled intercluster trunk (ICT) on Cisco CallManager. Cisco CallManager registers with the gatekeeper through the ICT, and CME registers with the gateke

[OSL | CCIE_Voice] QOS again: is cRTP supported with MLPPP

Hi List, is cRTP in combination with MLPPP supported? Where can i find some documenatation about this? I searched on CCO, but i did not find anything meaningful to me. If cRTP is supported, should i enable it in a MQC Class or under the virtual-template interface? Any hints are welcome! /Rob

[OSL | CCIE_Voice] Antw: save/restore flash in GNS3

Hi, i used dynamips without gns3 and i dont know if there are any differences. But with dynamips you have to format the flash via the cli "format flash". After that yould should be able to write files to flash, as if you would use a real router. /Robert >>> Brian Valentine schrieb am Samstag,

[OSL | CCIE_Voice] Antw: Re: CCIE_Voice Digest, Vol 38, Issue 160

Oliver, in which way does the T.303 Timer influence the call-forwarding behavior? @Tanmay: Could you explain your requirement/task a litllte bit more? I do not know what you are trying to achieve. /Robert >>> oliver rodrigues schrieb am Samstag, 25. April 2009 um 11:21 in Nachricht 38a849e4e0

[OSL | CCIE_Voice] Antw: Re: ringback as MoH

MoH for Callers in Queue: Place Call on Hold Step, Place call Off Hold Step; User Hold Audio Source Ringback when transfered to Agent: Ringback.wav as Network Hold Audio Source on CTI-Ports /Robert >> Chris Parker schrieb am Freitag, 13. März 2009 um 21:51 >> in Nachricht 4232655dc40f12fe0a517a

[OSL | CCIE_Voice] Antw: ringback as MoH

Hi Chris, you will find the ringback Files somewhere in the Prompt Folders under the wfavvid Folder. HTH /Robert >>> Chris Parker schrieb am Freitag, 13. März 2009 um 21:29 >>> in Nachricht 893e62af20e46a862acf2667dcd517ab: > Hello, > > I have been looking for a wav file that has ringback in

Re: [OSL | CCIE_Voice] Antw: CME

tate. Then assign template to your ephone and reset. > Rgds > Alex > > - Original Message - > From: "Robert Schuknecht" > To: ; > Sent: Monday, February 23, 2009 12:00 PM > Subject: [OSL | CCIE_Voice] Antw: CME > > >>I would only bind one

[OSL | CCIE_Voice] Antw: Re: SIP-Trunk: No DTMF

I tried the SIP-TRUNK another time, and it is still not working. Now i was calling from CME to CCM, and i am using only g711 and the software mtp. When the call is established (both phone off-hook) i am able to invoke supplementary services (hold/unhold) but from CCM to CME the DMTF is not worki

[OSL | CCIE_Voice] Antw: CME

I would only bind one single-line ephone-dn to the phone. /Robert >>> omar itani schrieb am Montag, 23. Februar 2009 um 08:31 in Nachricht 3e39d9c0a60c649e169eff7cad816c12: > WHAT IS THE COMMAND FOR BR2 PHONE 3 UNABLE INITIATE A CONFERENCE ? > > _

[OSL | CCIE_Voice] Antw: Re: B-Channel Maintenance not working properly

Vik, so what would you do in the real lab, mess around with the B-Channel Maintenance or set the B-Channel selection to Top-Down, if not explicitly asked to avoid glare and the Provider is using Top-Down also. /Robert >>> Vik Malhi schrieb am Sonntag, 22. Februar 2009 um >>> 18:33 in Nachrich

[OSL | CCIE_Voice] Antw: B-Channel Maintenance not working

repeat" method. > > Also, status polling has to be enabled for it to work properly. > > One other thing, is this a MGCP or H323 gateway config that you are working > with? > > Ken > > ------ >

[OSL | CCIE_Voice] B-Channel Maintenance not working properly

Hi List, during my last Remote-Rack Sessions i noticed that the B-Channel Maintenance Status Parameter is not workking properly. Always when i configured it and restartet the CCM Srevice and the Gateway itself, it is working for some calls. And suddenly the Gateway is trying to call out over th

[OSL | CCIE_Voice] Antw: ATA186 dot1q ot not?

Hi Chris, i configure the switchport for the ATA always as access-port. I would only configure the switchport for dot1q if i am asked to explicitly. /Robert >>> Chris Parker schrieb am Sonntag, 22. Februar 2009 um >>> 14:48 in Nachricht b74096f513403f0e877b6a80074dcdb5: > Should the port conne

[OSL | CCIE_Voice] Unity Voicemail Port Configuration

Hi List, lets assume we are required to configure Unity with 3 Ports + 1 MWI Port. Normally all Voicemail Ports are in a Line Group. Would you leave the MWI Voicemail Port also in the Line Group or would you exclude the MWI Voicemail Port from the Line Group? /Robert

[OSL | CCIE_Voice] SIP-Trunk: No DTMF

Hi List, last night i faced a problem with a SIP-Trunk from CCM to CCME. On the Call-Flow PHONE-1->CCM->SIP-Trunk->CCME->PHONE-2 i could only get the first DTMF-Tone, from Phone-1, through any subsequent DTMFs were not received by Phone-2. DTMF Tones from PHONE-2 to PHONE-1 were working just fi

Re: [OSL | CCIE_Voice] Antw: Re: Working on Unity Huntgroup scenario.

Ahhh, ok now i understand. Thanks for the answer. /Robert >>> Mark Snow schrieb am Donnerstag, 19. Februar 2009 um >>> 03:20 in Nachricht 8143166bd859658b76b9025149cae0c3: > CSS of the VM Pilot is simply the CSS that is applied to the Call > Coverage on the Phone Line when you click the Fwd [

Re: [OSL | CCIE_Voice] Antw: Re: Working on Unity Huntgroup scenario.

UPPS, i just tested this by myself and i changed the CSS of the VM-Pilot, and i restarted the CCM Services. But i am still able to get to Unity by pressing the Messages Button.. Maybe Vik and/or Mark can shed some light on this? /Robert >>> Robert Schuknecht schrieb am Mit

Re: [OSL | CCIE_Voice] Antw: Re: Working on Unity Hunt group scenario.

S without any access to > the internal partitions. > All the VM ports were in an "internal" partition. > When I dial the VM pilot, it shouldn't work based on my changes, but it > does. > > - Scott > > > > On Wed, Feb 18, 2009 at 1:36 PM, Robert Schukne

[OSL | CCIE_Voice] Antw: Re: Working on Unity Hunt group scenario.

Scott, as far as i know, the CSS of the VM-Pilot is used to reach the Hunt-Pilot of the Hunt-List/Line-Group of the VM-Ports. See the Voicemail-Pilot as a Speeddial to the Hunt-Pilot. /Robert >>> schrieb am Mittwoch, 18. Februar 2009 um 17:59 in Nachricht 884fa9c92a4d8b0e56bd24f2851093dc: >

[OSL | CCIE_Voice] Antw: Re: SRST-VM: Never seen before...

ly. > > Can you explain how your config work for cfw scenario? I didn't see any > manipulation regarding RDNIS. I may not understand your config properly. > > Thanks, > > james > >> Message: 1 >> Date: Tue, 17 Feb 2009 21:30:05 +0100 >> From: "

[OSL | CCIE_Voice] Antw: Re: SRST-VM: Never seen before...

Hi, i found an solution for CFWA under SRST. Under call-manager-fallback i configured voicemail 1408204, so that the called number will be 1408204 + DN of the phone on which the Message Button was pressed. On the Dial-Peer which points to HQ i configured a translation rule which translates

[OSL | CCIE_Voice] SRST-VM: Never seen before...

Hi List, i was working on getting Voicemail to work under SRST. I wanted to use the CTI-RP Solution. Under call-manager-fallback i configured: voicemail 140820425.. call-forward busy 140820425.. call-forward noan 140820425.. timeout 12 I also configured the required dial-peer : dial-peer voice

[OSL | CCIE_Voice] Antw: Re: MGCP IOS GW and G711 with 30msec sample size

e anything else. CCM is controlling codec selection, etc. It would > make sense that it would control that as well. > > Cliff > > > > - Original Message - > From: "Robert Schuknecht" > To: "OSL CCIE Voice Lab Exam" > Sent: Saturday, Fe

[OSL | CCIE_Voice] MGCP IOS GW and G711 with 30msec sample size

Hi List, i have a scenario where i am asked to configure the G711 codec with 30msec sample size. For CCM and H323 Gateways it`s no problem. But for an MGCP IOS Gateway i don`t find the right commands to configure it. Has anybody any Tipp for this? /Robert

Re: [OSL | CCIE_Voice] Antw: Re: IPCCx not able to create CTI Porton CCM

Hi list, what if the jtapi client version is not in sync between CCM and IPCCx, could this cause the Problem? /Robert >>> Vik Malhi schrieb am Freitag, 13. Februar 2009 um >>> 05:36 in Nachricht 4b0fc346acebf5c81b66ec3b8cb777ac: > Using a Device Pool that doesn't exist. So if you renamed Defau

Re: [OSL | CCIE_Voice] Antw: Re: IPCCx not able to create CTI Porton CCM

Hi Vik and all, thanks for the replies! Ok the Device-Pool is another thing to look at. Could there be a problem with DC-Directory, also? If so, where do i have to look at? I am asking all that stuff, because during working on a lab (build from a colleague of mine), i faced the problem to not b

[OSL | CCIE_Voice] Antw: aar

Do you mean you want to see a 4-digit calling no when AAR kicks in? Go to your corresponding RouteList and set "Use Calling Partys External PhoneNumberMask" to off. HTH /Robert >>> hasan khalife schrieb am Donnerstag, 12. Februar 2009 um 22:40 in Nachricht 1adf8130f7c68e654ce5cea156bde4a9: > h

[OSL | CCIE_Voice] Antw: Re: IPCCx not able to create CTI Port on CCM

set for rmjtapi is it working? > > Best regards > Daniel Sobrinho > -- > From: "Robert Schuknecht" > Sent: Thursday, February 12, 2009 6:47 PM > To: "OSL CCIE Voice Lab Exam" > Subject: [OSL | CCIE_Voice] IPCCx not able to crea

[OSL | CCIE_Voice] Antw: Re: IPCCx not able to create CTI Port on CCM

n I've found is to reinstall at that > point. > > > ----- Original Message - > From: "Robert Schuknecht" > To: "OSL CCIE Voice Lab Exam" > Sent: Thursday, February 12, 2009 3:47 PM > Subject: [OSL | CCIE_Voice] IPCCx not able to create CTI Port

[OSL | CCIE_Voice] IPCCx not able to create CTI Port on CCM

Hi List, i am looking for possible causes that will IPCCx prevent from creating CTI Ports in CCM. The only cause i know of is the AXL Writes Service Parameter for the Database Layer Monitor. But there must be other things other than that. Any Tipp or hint is very welcome! /Robert

[OSL | CCIE_Voice] Antw: Re: Lab Scenarios questions

14:40 in Nachricht 9539453ca7a5b7335750098a8795640f: > Ok. So it's DNIS outbound requiring 3 to 11. I was thinking this was just > affecting inbound traffic. > > So what gets it set up 3 seconds faster? Or is that not possible? > > > - Original Message - >

[OSL | CCIE_Voice] Antw: Lab Scenarios questions

Cliff, not all the numbers you will dial wil have 10 or 11 digits. Think about your Emergency numbers Especially for the Emergency numbers you will have your router dial out as fast as possible. HTH /Robert >>> Cliff McGlamry schrieb am Dienstag, 10. Februar 2009 um 07:15 in Nachricht 803

[OSL | CCIE_Voice] Antw: SIP Failover commands

Ryan, some time ago i found an blog article from Vik, where he explained your scenario. For SIP failover you need the following: sip-ua retry invite 2 timers trying 400 You will find Viks article at this link: http://malhi.net/blog/?p=31 HTH /Robert >>> Ryan Trauernicht schrieb am Donnerstag

Re: [OSL | CCIE_Voice] Antw: Re: Fw: Re: VPIM between CUEand Unityusing IPaddresses --SOLVED --

send an Message from the host unity.ccie-voice.local and sender of @ccie-voice.lab. In turn CUE rejected this message because the 2 domains did not match. Thanks for your replies! /Robert >>> Robert Schuknecht schrieb am Dienstag, 27. Januar 2009 >>> um

Re: [OSL | CCIE_Voice] Antw: Re: Fw: Re: VPIM between CUE and Unityusing IPaddresses

I have also some problems to get VPIM work, using IP-Addresses. Message from CUE to Unity work! Message from Unity to CUE does not working: - On CUE i see in the debugs that Unity tries to send a Message to CUE, but CUE rejects it - Here is the debug output: 3567 01/27 23:10:41.798 netw smtp

Re: [OSL | CCIE_Voice] Wtrlt: RE: Antw: BACD big trouble , anybody did thisbe4? -Partially working -

27 11:44:16.330: //302//MSW :/msw_synth_get_stream_state: genericStream > 0x47AA54EC is in state MSW_S_IDLE > > Jan 27 11:44:16.334: //302//MSM :/ms_associateDone: First Buf Play at > 04:06:44.068 of en_bacd_options_menu.au > > When CME phone calls B-ACD there is no trans

[OSL | CCIE_Voice] Antw: Re: Fw: Re: VPIM between CUE andUnity using IPaddresses

741 01/27 09:29:10.787 netw smtp 6 MAIL FROM:: RCPT TO:<2...@ccie-voice.lab> 3741 01/27 09:29:10.788 netw smtp 5 553 5.1.2 3741 01/27 09:29:10.792 netw smtp 6 RCPT TO:: RSET 3741 01/27 09:29:10.799 netw smtp 6 RSET: QUIT 3741 01/27 09:29:10.800 netw smtp 5 221 closing channel >>> R

[OSL | CCIE_Voice] Antw: Re: Fw: Re: VPIM between CUE and Unity using IPaddresses

Hi, i think i need some more hints or help. In my opinion i have configured the system as discussed in this threat, but i cant send vpim messages from CUE to Unity, i always get "550 5.7.1 Unable to relay for 2...@192.168.120.133 " (192.168.120.133: IP of Unity) Any ideas? /Robert >>> S

[OSL | CCIE_Voice] Wtrlt: RE: Antw: BACD big trouble , anybody did thisbe4? -Partially working -

2400 ext 226 > > Email: dmann...@me.vccs.edu > www.mecc.edu > > > > -Original Message- > From: ccie_voice-boun...@onlinestudylist.com > [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Robert > Schuknecht > Sent: Monday, January 26, 2009 2:09

[OSL | CCIE_Voice] Antw: BACD big trouble , anybody did this be4? -Partially working -

Hi, i have just tried it with SIP Dial-Peers instead of H323 and it works partially. Call from CME-Phone to BACD: working OK Call from CCM Phone via GK and G729: partially working When i call from CCM i cant hear the first Prompt (Drop Through Prompt) but i am able to hear the second Prompt (Al

[OSL | CCIE_Voice] Antw: Re: Fw: Re: VPIM between CUE and Unity using IPaddresses

Hi all, as far as i know (i have only read about it) it is not necessary, to create a DNS Host Record fuer CUE. IF you enter the IP-Address, with square Bracktes, in the Smart Host Filed. Example Smart Host: [192.168.122.130]. I will try this out, by myself, during my next learning session, tom

[OSL | CCIE_Voice] Antw: Re: GK trunk h323 id for pub and sub selection

Hi, here is an old post of mine: On Sat, Jan 17, 2009 at 9:53 PM, Robert Schuknecht wrote: > Hi, > > the _1 or _2 represents the CTI-ID of the CCMs. The only way to change > them, which i know off, is to change the values in the SQL Database. But i > think thats not supported

[OSL | CCIE_Voice] Antw: Thoughts on SIP Trunk with multicast MOH

Hi, i recently have found a link to a blog of Vik Mahli. There he explains some capabilities of sip-trunks: http://malhi.net/blog/?p=31 Regarding his blog, multicast Music on Hold is not supported on sip trunks HTH /Robert >>> Ryan Trauernicht schrieb am Montag, 12. Januar >>> 2009 um 14:53

[OSL | CCIE_Voice] Antw: Outbound PSTN calls using Same Number

Hi, then you could use "clid network-number", under the dial-peer. /Robert >>> kamal yousaf schrieb am Dienstag, 30. Dezember >>> 2008 um 14:28 in Nachricht c9a66d34e8c459d0752610cf9d017863: > Hi , > > If we are required to configure CME such that all outbound calls through > PSTN use same E

[OSL | CCIE_Voice] Antw: CAC mechanism to limit the number of calls overthe WAN

Robertico, the calculation you have used is for Traffic Shaping/LLQ on the Routers. The Callmanager does not know about L2-Header Overhead. Callmanager, for CAC, uses 80kbit/s for G711 and 24 kbit/s for G729 calls. HTH /Robert >>> Robertico Gonzalez schrieb am Donnerstag, >>> 11. Dezember 200

[OSL | CCIE_Voice] Antw: Re: H323 with SRST: Block incoming calls tospecificDN

CCMSUB" > destination-pattern [12]... > preference 1 > session target ipv4:10.1.200.20 > dtmf-relay h245-alpha > ! > Cheers > Alex > > > - Original Message - > From: "Vik Malhi" <[EMAIL PROTECTED]> > To: "Alex" <[EMAI

Re: [OSL | CCIE_Voice] Antw: Re: H323 with SRST: Block incomingcalls tospecificDN

Oh, you are correct! This won´t work, because the blocking is active all the time So, the only way, i can think of right now, is something like my solution in an earlier post. I am/was hoping that there is some easier/faster way Right now, i cant do any further testing, cause it´s almo

[OSL | CCIE_Voice] CME: Phone should be able to dial only by pressing New Call

Hi list, imagine you are asked to configure a CME-Phone in such a way that it´s only possible to make a new call by pressing the "New Call" softkey. How could this be achieved? I tried it with the "no auto-line" under ephone. The BAckside of this is, on incoming calls, i have to press the "Ans

[OSL | CCIE_Voice] Antw: Re: H323 with SRST: Block incoming calls tospecificDN

Many thanks for the answer. I think i will use the solution from Mark Cardwell. He suggested to use the "call-block" Dial-Peer Command. It seems that this is the fastest way to fullfill all the requirements (H323 GW; 2003 ONLY not reachable, from intl numbers, in SRST-Mode). /Robert >>> Vik

Re: [OSL | CCIE_Voice] Antw: RE: H323 with SRST: Block incoming callsto specificDN

Drive, Suite 100, Greenbelt, MD 20770 | > mailto:[EMAIL PROTECTED] > D: 571.225.0132 | http://www.presidio.com/ > > > -Original Message- > > From: Robert Schuknecht [mailto:[EMAIL PROTECTED] > Sent: Tuesday, November 11, 2008 4:39 PM > To: Cardwell,

[OSL | CCIE_Voice] Antw: H323 with SRST: Block incoming calls to specificDN !!! GOT IT !!!

codec 1 voice-class h323 1 incoming called-number A2003$ dtmf-relay h245-alphanumeric ip qos dscp cs3 signaling no vad Hope this helps someone, or at least, points in the right direction. /Robert >>> Robert Schuknecht<[EMAIL PROTECTED]> schrieb am Dienstag, 11. November 20

[OSL | CCIE_Voice] Antw: Re: H323 with SRST: Block incoming calls to specific DN

locking just calls from intl #'s. > > But an inbound voice translation profile on your voice-port could easily see > 2003 being targeted as DNIS and strip it. That would cause inbound calls to > 2003 to go to a fast busy. > > > > On Tue, Nov 11, 2008 at 3:47 PM,

[OSL | CCIE_Voice] H323 with SRST: Block incoming calls to specific DN

Hi, i am totally stuck with the following scenario: 1) H323 Gateway with SRST 2) when SRST is off #2003 is able to call everywhere 3) when SRST is off #2003 can be called from everywhere 4) when SRST is active #2003 should not called from international numbers / or from all numbers 5) when SRST

[OSL | CCIE_Voice] CME: New call only via Softkey, not via speaker button

Hi List, i have the following scenario: On CME i am only allowed to make a new call via the "New-Call" Softkey. The Speaker Button should not work. I am out of ideas, at the moment. /Robert

[OSL | CCIE_Voice] Antw: IPCCX ring back tone to caller

Erick, i did not try the following, but what if you make the MOH Audio Source, of the IPCC CTI Ports to play a *.wav file with Ring-Tone. Search on Publisher/IPCCx under: c:\program files\wfavvid..., for *.wav files i think you will find some adequat soundfiles. HTH /Robert >>> Erick Pineda<

[OSL | CCIE_Voice] Only a few days left.... Need help on redundant IP-Phone Services!!!!

Hi List, i have only a few days left until my first real LAB-Exam (its on Friday). And i don´t get the IP.Phone Services run in a redundant fashion. I would like to use DNS SRV Records, for services like IPMA, EM,FastDials, Addressbook...etc. But i can´t get it to work. Could anybody please exp

[OSL | CCIE_Voice] Antw: VMPI problem.stucked for hours and noprogress.who knows VMPI and exchange 2K well may help.

Jeremy, if i understand your debug right, then are not using Fully Qualified Domain Names. In the trace i saw the following line: "netw smtp 5 RCPT TO [EMAIL PROTECTED]" . Are you using ccie as FQDN? If yes, try to change it to ccie.lab or any other Top-Level Domain which your DNS Server is us

[OSL | CCIE_Voice] Antw: RE: Voicemail Access when AAR kicks in

dio Networked Solutions > 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | > mailto:[EMAIL PROTECTED] > D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ > > > -Original Message- > > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On

[OSL | CCIE_Voice] Antw: H323 GW PSTN -> BR1 call.

James, in my opinion, you have two options: 1) Configure your H323 Gateway, with the IP-Address of the Interface pointing to Callmanager, in the Gteway Config Page on Callmanager 2) Don´t issue the "gateway" command on the RTR. Without the "gateway" command the RTR won´t register to any gateke

[OSL | CCIE_Voice] Voicemail Access when AAR kicks in

Hi, should Voicemail Access, from BR1, work when AAR kicks in? In my LAB it is currently not working and i don´t know if it works like desinged or if i made an silly mistake. Normal Calls from BR1 to HQ are rerouted over PSTN, only Voicemail calls are not. Regards /Robert

[OSL | CCIE_Voice] Antw: POD:18 - gatekeeper section --> task5

Deniz, it looks like that the PSTN-WAN has no Dial-Peer to your dialed number. Is your BR2 PSTN Connection working? Remember, the call flow will be: CCM->GK->Remote GK/PSTN-WAN->E1-PRI/E1-R2->BR2. So, when your BR2 E1 ist not up & running then the desired Dial-Peer, on the PSTN-WAN Router, will

[OSL | CCIE_Voice] Antw: IPCC lab Gotcha

Steve, were your CTI-Ports registered? Or, was only the CTI-Route-Point resgistered? Maybe they have tweaked the Service-Parameters, so only one device was able to register. /Robert >>> Steve ccietester<[EMAIL PROTECTED]> schrieb am Dienstag, 4. November 2008 um 07:57 in Nachricht 6c50a8dd179a

[OSL | CCIE_Voice] Antw: Re: Gateway Channel selection control ???

Mark and all, could you please explain how to configure multiple groups of timeslots, on one controller? I have tried to configure it, but as soon as i want to configure the second time-slot group i get the following error: %A ds0-group was configured already. Please remove it to configure a pr

[OSL | CCIE_Voice] Antw: Problem with PVDM-2 Conference registration

Hi Michael, Hardware Conferencing is not supported with CCME 3.3. When you want to use a Hardware Conferencebridge with CCME you have to use CCME Version 4.0 or maybe higher, i don´t remember it at the moment. HTH /Robert >>> Michael Shavrov<[EMAIL PROTECTED]> schrieb am Mittwoch, 22. Oktober

Re: [OSL | CCIE_Voice] Antw: Volume 1 Task 16.5 on POD 11: VPIM beween Unityand CUE -RESOLVED-

're at it :) So vpim integration > requires Unity and CUE to be in different domains? Can it be made to work on > the same domain with different subdomains? > > > -Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Robert

[OSL | CCIE_Voice] Antw: Volume 1 Task 16.5 on POD 11: VPIM beween Unityand CUE -RESOLVED-

Hi, i found my own error. I did not place Unity/CUE in their own different DNS Domains. I had Unity in "voip.lab" and CUE in "cue.voip.lab". I changed CUE to "br2.lab" and all is working fine! /Robert >>> Robert Schuknecht<[EMAIL PROTECTED]> sc

[OSL | CCIE_Voice] Volume 1 Task 16.5 on POD 11: VPIM beween Unity and CUE

Hi, right know i trying to get VPIM to work. But i can´t figure out what i am doing wrong. I have configured VPIM regarding the Proctor Guide, but VPIM wont work. I try to send a messag from CUE to Unity, below you can see the CUE Debug: 4642 10/20 08:49:11.903 netw smtp 4 4642 10/20 08:49:11.9

[OSL | CCIE_Voice] CCM 4.1 and AD integration

Hi List, i am wondering if i should be prepared for doing an AD Integration in the actual LAB. What do you, especially those who have taken the lab, think? Should i expect to do an Integration? I dont want anybody to break the NDA, i only want to know what all the others are thinking about th

[OSL | CCIE_Voice] Antw: Re: Mapping video traffice

; > On Wed, Oct 8, 2008 at 12:55 AM, Robert Schuknecht <[EMAIL PROTECTED]>wrote: > >> Chris, >> >> for the class-map you will have to know the IP-Addresses and Port Numbers, >> which the Video Endpoints will use. So you can match an Access-List, which >> d

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