Re: [OSL | CCIE_Voice] RE : Mobile voice access a sking userid rather then pin
There is a CCM Service parameter to check calling no for MVA. You may set it to partial match to avoid issues with not having exact digit by digit match. See if it helps. --- On Thu, 6/24/10, naoufal.kerboute naoufal.kerbo...@cbi.ma wrote: From: naoufal.kerboute naoufal.kerbo...@cbi.ma Subject: [OSL | CCIE_Voice] RE : Mobile voice access asking userid rather then pin To: laurent.bourm...@orange-ftgroup.com, . ccie_voice@onlinestudylist.com Date: Thursday, June 24, 2010, 10:04 PM check the caller id coming from pstn to UCM and youe remote destination number. May be you need to apply any transformation in the calling number to match the remote destination number Message d'origine De: ccie_voice-boun...@onlinestudylist.com de la part de laurent.bourm...@orange-ftgroup.com Date: jeu. 24/06/2010 18:30 À: . Objet : [OSL | CCIE_Voice] Mobile voice access asking userid rather then pin Hi, It seems that I have an issue getting the MVA oerational. Actually I use a H323 gatway connected to the CUCM, so when I call from my remote destination I have the prompt saying Please enter your userid ... rather than getting the pin prompt. I did a trace on the CUCM then it seems that my remote destination is properly recognized : 2010-06-24 18:21:01,269 DEBUG [http-8080-Processor24] controller.IVRGetAudioFile - [CCM_IVR]:: IVR get Filename 2010-06-24 18:21:01,270 DEBUG [http-8080-Processor24] controller.IVRGetAudioFile - [CCM_IVR]::getLocaleFile() 2010-06-24 18:21:01,271 DEBUG [http-8080-Processor24] controller.IVRGetAudioFile - [CCM_IVR]::getLocaleFile() fileName = 1.au and locale = en_US 2010-06-24 18:21:01,271 DEBUG [http-8080-Processor23] controller.IVRCalleridLookup - [CCM_IVR]::execute(): action start 2010-06-24 18:21:01,272 DEBUG [http-8080-Processor24] controller.IVRGetAudioFile - [CCM_IVR]::getLocaleFile() fileName = 1.au and locale = en_US 2010-06-24 18:21:01,273 DEBUG [http-8080-Processor23] controller.IVRCalleridLookup - [CCM_IVR]::srcdir: en_US 2010-06-24 18:21:01,274 DEBUG [http-8080-Processor23] controller.IVRCalleridLookup - [CCM_IVR]::-new-code-got-callerid-as-remotedest: +447976852817 2010-06-24 18:21:01,275 DEBUG [http-8080-Processor24] controller.IVRGetAudioFile - [CCM_IVR]:: getting the file now 2010-06-24 18:21:01,275 DEBUG [http-8080-Processor23] controller.IVRCalleridLookup - [CCM_IVR]::srcdir: en_US sessionid:1472FAADA95E59CFE334AD38AFA395FF 2010-06-24 18:21:01,277 DEBUG [http-8080-Processor23] controller.IVRCalleridLookup - [CCM_IVR]::remotedest: +447976852817 sessionid:1472FAADA95E59CFE334AD38AFA395FF 2010-06-24 18:21:01,278 DEBUG [http-8080-Processor23] controller.IVRCalleridLookup - [CCM_IVR]::Host: 10.100.210.11:8080 2010-06-24 18:21:01,279 DEBUG [http-8080-Processor23] controller.IVRCalleridLookup - [CCM_IVR]::Content-Type application/x-www-form-urlencoded 2010-06-24 18:21:01,279 DEBUG [http-8080-Processor23] controller.IVRCalleridLookup - [CCM_IVR]::Connection: close 2010-06-24 18:21:01,280 DEBUG [http-8080-Processor23] controller.IVRCalleridLookup - [CCM_IVR]::Accept: text/vxml, text/x-vxml, application/vxml, application/x-vxml, application/voicexml, application/x-voicexml, text/plain, text/html, audio/basic, audio/wav, multipart/form-data, application/octet-stream 2010-06-24 18:21:01,280 DEBUG [http-8080-Processor23] controller.IVRCalleridLookup - [CCM_IVR]::User-Agent: Cisco-IOS-C3845/12.4 2010-06-24 18:21:01,281 DEBUG [http-8080-Processor23] util.IVRDBInterface - [CCM_IVR]--- getUserIdFromRemoteDestination- DB Query -- remoteDest = +447976852817 2010-06-24 18:21:01,285 DEBUG [http-8080-Processor23] util.IVRDBInterface - [CCM_IVR]--- isPartialMatchEnabled --DB Query -- sql: = select paramValue from processconfig where paramName ='RemDestCallerIDMatchType' and tkService ='0' 2010-06-24 18:21:01,325 DEBUG [http-8080-Processor23] util.IVRDBInterface - [CCM_IVR]--- isPartialMatchEnabled --DB Query -- RemDestCallerIDMatchType: = 1 2010-06-24 18:21:01,327 DEBUG [http-8080-Processor23] util.IVRDBInterface - [CCM_IVR]--- getUserIdFromRemoteDestination() -- partial match enabled 2010-06-24 18:21:01,328 DEBUG [http-8080-Processor23] util.IVRDBInterface - [CCM_IVR]--- getNumDigitsPartialMatch --DB Query -- sql: = select paramValue from processconfig where paramName ='RemDestCallerIDMatchDigits' and tkService ='0' 2010-06-24 18:21:01,329 DEBUG [http-8080-Processor23] util.IVRDBInterface - [CCM_IVR]IVRDBInterface.getUserIdFromRemoteDestination() --java.lang.NullPointerException 2010-06-24 18:21:01,329 DEBUG [http-8080-Processor23] controller.IVRCalleridLookup - [CCM_IVR]::ccmusername: null 2010-06-24 18:21:01,330 DEBUG [http-8080-Processor23] controller.IVRCalleridLookup - [CCM_IVR]::ccmusername:is Null ?? null 2010-06-24 18:21:01,331 DEBUG [http-8080-Processor23] controller.IVRCalleridLookup - [CCM_IVR]IVRCallerIdLookup::execute(): ccmusername is Null and action forwarding IVRUserid 2010-06-24
[OSL | CCIE_Voice] vouchers
Hi List, I've discussed this with PL team and taken their permission before posting this. I've around 20 odd vouchers available at minimal price. Those are left overs after passing my lab. If anyone interested pl PM me. All vouchers are valid for V3. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] V3 Attempt Two
I cracked mine on 3rd go in V2. Be consistent...good luck --- On Wed, 9/2/09, Ravindra Lakpriya lakpr...@gmail.com wrote: From: Ravindra Lakpriya lakpr...@gmail.com Subject: Re: [OSL | CCIE_Voice] V3 Attempt Two To: Tanner Ezell tanner.ez...@gmail.com Cc: ccie_voice@onlinestudylist.com Date: Wednesday, September 2, 2009, 11:00 AM Let's hope for the best man. U ll nail it. All the best dude. On Wednesday, September 2, 2009, Tanner Ezell tanner.ez...@gmail.com wrote: Good luck Jonathan, look forward to hearing the results! On Tue, Sep 1, 2009 at 9:53 PM, Jonathan Charles jonv...@gmail.com wrote: OK, took v3 again in RTP today... finished 30 minutes early... Well, not really... what happened was that I was doing some last minute tweaking (just retesting stuff, cleaning up some config) and some key huge point items stopped working... I undid what I did to break stuff, got up and walked away... yes, there was 30 minutes on the table, but it could have been the death of me... Anyway, waiting on results I would like to claim optimism, as I studied the crap out of my shortcomings last time, but I have done this before where I walked out of a lab pretty confident to see zero on sections I thought I aced... to be honest, I am like 85% sure I failed again. As they all say, the test is fair, nothing out of left field, some surprises on what was on there and what wasn't... there are some sleazy traps, but if you have a clue, you will work around em pretty quick... Took the first one in SJ, took this one in RTP... so, I can compare... In SJ, the phones are nailed to the walls in the cubicle... in RTP, they are on the desk (so you can flip em over and look at em...)... not sure which I prefer... I kinda like throwing them at the wall... But then again, in SJ, Ben Ng is sitting 4 feet from you, so, no intimidation there... I saw the remnants of the old v2 labs sitting in RTP, still had phones and fax machines... looked abandoned... Everything else I could say would be NDA... so, guys, do what you always do, look for the flurry of questions on 'how do I in this group or as veiled customer issues on Puck As a joke, here are the four questions I would ask: Why on this day are we limited on how we can dial? When on all other days we can dial however we want? Why on this day must we use frame-relay, when on all other days, all of our customers have MPLS? Why on this day are we running unpatched, basically beta-versions of CUCM, CUPS, CUCCX, when on other days we can install patches to get around bugs? Why on this day do I have to fly all the way to Raleigh and start the test at 7:15AM, when the guys who go to San Jose get to sleep in and take their test at 9:00? Jonathan If you are Jewish, those are funny. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Ravindra Lakpriya +94 773 532 094 ___ 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
Re: [OSL | CCIE_Voice] Thanks
Hi List, The journey I started last year with a CCIE Voice boot-camp from a freelancer. The boot camp was little frustrating and the only good thing was that the instructor introduced me to the PL and OSL. During my earlier attempt I practiced on a small Dynamips/VM Ware simulated home-lab in addition to PL Remote racks. I also got hold of some hardware and built a rack for practice. Covering the technology was not a problem in the lab, but there was no defined strategy to attack. In the lab exam, I started covering section by section. By the end of the lab it was whole mess in my mind and fixing even smallest of issues were taking longer then expected. I didn't have the testing strategy as well. I couldn't pass that time, not even close to passing. This time I worked a lot on the test taking and testing strategy both. My recommendation in addition to covering the technology is to work on the attack order you are going to follow in the lab, Atleast that was the problem for me. Also don't forget to have a workable testing strategy in place, that could be the difference between pass/fail. I would also recommend to go through the Online Voice VLectures available at http://www.ipexpert.com/index.cfm/a/p/vlectures. There is lot of relevant information available in those vlectures. Wish good luck to all, Thanks, Kapil Atrish --- On Mon, 6/29/09, Cristobal Priego cristobalpri...@gmail.com wrote: From: Cristobal Priego cristobalpri...@gmail.com Subject: Re: [OSL | CCIE_Voice] Thanks To: kapil atrish nice_cha...@yahoo.com Cc: ccie_voice@onlinestudylist.com Date: Monday, June 29, 2009, 6:04 AM congratulations could you share your expirience with us what tips could you provide? 2009/6/28 kapil atrish nice_cha...@yahoo.com Hi list, I've passed the lab exam and would like to thanks each one of you for your contributions to this list. I would also like to extend my thanks to PL After-hour support team for providing the instant support on various issues I faced during practice sessions. Thanks, Kapil Atrish CCIE Voice# 24706
[OSL | CCIE_Voice] Thanks
Hi list, I've passed the lab exam and would like to thanks each one of you for your contributions to this list. I would also like to extend my thanks to PL After-hour support team for providing the instant support on various issues I faced during practice sessions. Thanks, Kapil Atrish CCIE Voice# 24706
Re: [OSL | CCIE_Voice] GDM configuration with notification on2phones
xlate the mwi to say . Create two ephone-dns: ephone-dn 1 number label 3101 ! ephone-dn 2 number label 3102 ! Apply ephone-dn's using regular overlay option, keep in mind to put dn 1 and 2 as first option against respective buttons. That way, you'll meet the phone display requirement and still get the mwi envelope...Don't have lab to test it, but not sure why it won't work. --- On Tue, 6/23/09, Michael Ciarfello mciarfe...@iplogic.com wrote: From: Michael Ciarfello mciarfe...@iplogic.com Subject: RE: [OSL | CCIE_Voice] GDM configuration with notification on2phones To: kapil atrish nice_cha...@yahoo.com, Cristi Radescu cristian.rade...@crescendo.ro Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Tuesday, June 23, 2009, 7:50 AM What about 3101 requirement? From: kapil atrish [mailto:nice_cha...@yahoo.com] Sent: Friday, June 12, 2009 4:50 AM To: Cristi Radescu; Michael Ciarfello Subject: Re: [OSL | CCIE_Voice] GDM configuration with notification on2phones Yes, its possible. I've tried it and it works. Assume u want the envelope on 3102 which is a secondary line on the phone and mwi requests are coming for 3100. voice translation-rule 1 rule 1 /^\(80003100\)/ /80003102/ rule 2 /^\(80013100\)/ /80013102/ ! voice translation-profile mwi translate called 1 ! dial-peer voice 123 voip session protocol sipv2 session target ipv4:CUE_IP_ADD destination-patt VM_PILOT incoming called-nu 800[0-1]... translation-prof outgoing mwi codec g711 no vad dtmf-relay sip-notify ! The MWI will get Xlated to 3102, and since its on line 2, u'll get only the envelope and not the Light. --- On Thu, 6/11/09, Michael Ciarfello mciarfe...@iplogic.com wrote: From: Michael Ciarfello mciarfe...@iplogic.com Subject: Re: [OSL | CCIE_Voice] GDM configuration with notification on2phones To: Cliff McGlamry cl...@mcglamry.net, Cristi Radescu cristian.rade...@crescendo.ro, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Thursday, June 11, 2009, 8:03 AM Well, we are thinking and using our experience to try things. The CCIE is also a journey. I didn't pass back in Feb, but I came out a lot sharper on the skills and basic building blocks. Even these no solution scenarios are valuable because you never know what a customer will ask for and sometimes the answer is NO. Wouldn't that be something to put no win scenarios on the lab and you have to explain why it doesn't work? I'll shut up now before someone thinks it's a good idea. lol. From: Cliff McGlamry [cl...@mcglamry.net] Sent: Wednesday, June 10, 2009 11:26 AM To: Michael Ciarfello; Cristi Radescu; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] GDM configuration with notification on2phones My understanding is that you are correct. The answer thus far is no. From: Michael Ciarfello Sent: Tuesday, June 09, 2009 10:44 PM To: Cliff McGlamry ; Cristi Radescu ; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] GDM configuration with notification on2phones Did this ever get solved? Am I correct in saying the answer so far is NO? From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cliff McGlamry [cl...@mcglamry.net] Sent: Thursday, May 28, 2009 3:47 PM To: Cristi Radescu; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] GDM configuration with notification on 2phones Actually, you might be able to do it if it is on CME. Assign button 2 as an overlay, and put the mailbox number DN on the overlay ephone-dn. It should be hard forwarded so it will never ring, but I bet that would make the envelope appear the way being discussed. From: Cristi Radescu Sent: Thursday, May 28, 2009 5:22 AM To: 'ccie_voice@onlinestudylist.com' Subject: Re: [OSL | CCIE_Voice] GDM configuration with notification on 2phones I think this is not possible. With „secondary number” or „overlay” it will not work. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of David Corbeil Sent: 27 May, 2009 8:42 PM To: 'ccie_voice@onlinestudylist.com' Subject: [OSL | CCIE_Voice] GDM configuration with notification on 2 phones Hi, I want to know if it’s possible to have the voicemail letter on the second line of 2 phones without changing the phone facing. Example: Phone 1 Line1 : 3001 Line2 : 3101
Re: [OSL | CCIE_Voice] Problem with outgoing calls from Branch via T1 Pri to simulated PSTN router
That's a common issue. Create one DP on CME as following: dial-peer voice xxx pots incoming called-nu . direct-inward-dial ! --- On Thu, 5/28/09, ccieid1ot ccieid...@gmail.com wrote: From: ccieid1ot ccieid...@gmail.com Subject: Re: [OSL | CCIE_Voice] Problem with outgoing calls from Branch via T1 Pri to simulated PSTN router To: Padmanabhan, Padhu pa...@ti.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Thursday, May 28, 2009, 10:20 AM Just add another dial-peer for 911 with forward-digt all. On Wed, May 27, 2009 at 5:09 PM, Padmanabhan, Padhu pa...@ti.com wrote: Hi, I am just getting started with my voice lab preps…I have a branch router connected via WAN as well via T-1(crossover) to PSTN simulator running CME. I am unable to call simulated phones lines on the CME using the T-1 pri from branch. As soon as I dial 911 it goes into connected state from Branch-1. However from the CME /PSTN I can call inbound to the BR1 phones. Pasted below is relevant config. BR1 router configured as h323 gw and added to RG-RL-RP. Pattern 911 added to partition and included in the CSS for the BR1 phone. Any ideas?. Thanks,Padhu Branch-1: isdn switch-type primary-ni voice class h323 1 h225 timeout tcp establish 3 controller T1 1/0/0 pri-group timeslots 1-24 description connection from BR1 to PSTN interface Serial1/0/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice no cdp enable dial-peer voice 999 pots description Calls to PSTN using Local gateway destination-pattern .T incoming called-number . direct-inward-dial port 1/0/0:23 ! dial-peer voice 1 voip voice-class h323 1 incoming called-number . dtmf-relay h245-alphanumeric ! dial-peer voice 775 voip destination-pattern 775255 voice-class h323 1 session target ipv4:10.1.40.10 pstn-sim: isdn switch-type primary-ni isdn gateway-max-interworking voice class h323 1 h225 timeout tcp establish 3 controller T1 0/0 framing esf clock source internal linecode b8zs cablelength short 133 pri-group timeslots 1-24 description Connection to BR1 interface Serial0/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn protocol-emulate network isdn incoming-voice voice no cdp enable dial-peer voice 775 pots description calls to BR1 destination-pattern 775255 incoming called-number . port 0/0:23 forward-digits all telephony-service max-ephones 8 max-dn 30 ip source-address 10.1.99.1 port 2000 system message CME as PSTN Simulator max-conferences 8 gain -6 transfer-system full-consult create cnf-files version-stamp Jan 01 2002 00:00:00 ! ! ephone-dn 1 number 911 label Emergency_911 ephone 1 device-security-mode none mac-address 0002.FD65.9D5B button 1:1 2:2 3:3 4:4 button 5:5 6:6
Re: [OSL | CCIE_Voice] ATA IVR not responding
Hi Michael,
Can u give more inputs reg turning IVR off via config-register? Haven't heard
abt this before, sounds interesting.
Yes the Red light does come on and goes off perfectly fine. I was able to
make/receive calls to/from ata phone.
ATA is not reachable now can't change the software image...:(
I did shake it, opened the box and tried pressing the little button but no
good.:(
--- On Sun, 5/3/09, Michael Ciarfello mciarfe...@iplogic.com wrote:
From: Michael Ciarfello mciarfe...@iplogic.com
Subject: RE: [OSL | CCIE_Voice] ATA IVR not responding
To: kapil atrish nice_cha...@yahoo.com, ccie_voice@onlinestudylist.com
ccie_voice@onlinestudylist.com
Date: Sunday, May 3, 2009, 9:00 AM
#yiv1591233982 P {
MARGIN-TOP:0px;MARGIN-BOTTOM:0px;}
Can you turn the IVR off via a config register? Might be disabled.
When you pick up the phone does the red light come on?
Try the factoryreset thingie.
Try re-loading the software image.
Try shaking it. (lol. just kidding)
From: ccie_voice-boun...@onlinestudylist.com
[ccie_voice-boun...@onlinestudylist.com] On Behalf Of kapil atrish
[nice_cha...@yahoo.com]
Sent: Saturday, May 02, 2009 11:31 PM
To: ccie_voice@onlinestudylist.com; Cliff McGlamry
Subject: Re: [OSL | CCIE_Voice] ATA IVR not responding
Yes, that's what I am trying to invoke the IVR but no response. I've changed
the cable, Phone, Phone port, removed lan cable and tried whole thing again but
no good.
--- On Sat, 5/2/09, Cliff McGlamry cl...@mcglamry.net wrote:
From: Cliff McGlamry cl...@mcglamry.net
Subject: Re: [OSL | CCIE_Voice] ATA IVR not responding
To: kapil atrish nice_cha...@yahoo.com, ccie_voice@onlinestudylist.com
Date: Saturday, May 2, 2009, 11:30 PM
Did you push the button on top of the ATA after picking up the phone on port
1? That's how you activate the IVR menu.
What exactly have you done?
- Original Message -
From:
kapil atrish
To:
ccie_voice@onlinestudylist.com
Sent: Saturday, May 02, 2009 8:25 AM
Subject: [OSL | CCIE_Voice] ATA IVR not responding
Hi list,
M not able to access the IVR menu on ATA.I know it should work on Phone 1 port,
no success on any of the port. Reboot didn't help. Removed lan cable and tried,
no success. Checked physical layer stuff. Has anyone had this issue earlier and
any troubleshooting
I can do??? M running SCCP image on it.
I was testing the ATA vlan stuff and put in a Vlan and OpFlag which made my ATA
unreachable. Now I don't have a switch to configure in the specific vlan and
access my ATA. Since IVR is not working I am not even able to revert the ATA
settings.
Thanks in advance...
Re: [OSL | CCIE_Voice] ATA IVR not responding
Yes, that's what I am trying to invoke the IVR but no response. I've changed the cable, Phone, Phone port, removed lan cable and tried whole thing again but no good. --- On Sat, 5/2/09, Cliff McGlamry cl...@mcglamry.net wrote: From: Cliff McGlamry cl...@mcglamry.net Subject: Re: [OSL | CCIE_Voice] ATA IVR not responding To: kapil atrish nice_cha...@yahoo.com, ccie_voice@onlinestudylist.com Date: Saturday, May 2, 2009, 11:30 PM Did you push the button on top of the ATA after picking up the phone on port 1? That's how you activate the IVR menu. What exactly have you done? - Original Message - From: kapil atrish To: ccie_voice@onlinestudylist.com Sent: Saturday, May 02, 2009 8:25 AM Subject: [OSL | CCIE_Voice] ATA IVR not responding Hi list, M not able to access the IVR menu on ATA.I know it should work on Phone 1 port, no success on any of the port. Reboot didn't help. Removed lan cable and tried, no success. Checked physical layer stuff. Has anyone had this issue earlier and any troubleshooting I can do??? M running SCCP image on it. I was testing the ATA vlan stuff and put in a Vlan and OpFlag which made my ATA unreachable. Now I don't have a switch to configure in the specific vlan and access my ATA. Since IVR is not working I am not even able to revert the ATA settings. Thanks in advance...
[OSL | CCIE_Voice] CCM 7 on VMWare
Hi list, I want to know if CCM 7 is supported on VMWare workstation 5.0? The hardware I've is AMD quad-core, 4gig ram. Will that work, if someone who has tested it can comment please? Thanks..
Re: [OSL | CCIE_Voice] Dial-peer overlapping
I've tested in on CCM and CME but over the PSTN. On CCM side: Create a Voice-Mail profile to_vm, select the Voice-mail pilot and put the external phone number mask as . Create a Route Point with DN 22xxx, do call-forward all to VM, select the VM profile to_vm. To make it work over the WAN, configure Translation Pattern to see the 1#.2 (saying Tech-prefix for both sides is 1#) or Called number mask to 2. On CME side: Create a Xlation-rule (say rule no 1) to Xlate 24xxx to 4xxx. Create ephone-dn 24xxx, do call-forward all to CUE. Create a Xlation proflile (say to_vm), translate called 1 translate redirected-called 1 ! on the CUE dial-peer, apply the Xlation profile to_vm in outbound direction. Additional Xlation rule to make it work over the WAN. --- On Tue, 3/31/09, Chris Parker cpar...@cparker.us wrote: From: Chris Parker cpar...@cparker.us Subject: Re: [OSL | CCIE_Voice] Dial-peer overlapping To: Duy Nguyen ccieid...@gmail.com Cc: ccie_voice@onlinestudylist.com CCIE_Voice@onlinestudylist.com Date: Tuesday, March 31, 2009, 4:24 PM Duy, OK I forgot that it is going to be an ephone dialing this number so the ITS is processing each digit in real time and will never let you hit the 5th digit once its made a match on the first 4. Chris Duy Nguyen wrote: Chris, It actually keeps hitting the Dial-peer voice 2300 voip because of a shorter pattern when I dial 2400. I was able to achieve this by making it more explicit. Allow to dial HQ phones ! dial-peer voice 2300 voip preference 1 destination-pattern [23][0-1].. session target ras tech-prefix 1 dtmf-relay h245-alphanumeric ip qos dscp cs3 signaling no vad ! Allow to go to HQ Phone's VM directly dial-peer voice 22000 voip preference 1 destination-pattern 22... session target ras tech-prefix 1 dtmf-relay h245-alphanumeric ip qos dscp cs3 signaling no vad ! Allow to hit the CME ph's VM directly. Also put in 24001 pattern into E.164 in CUE. ephone-dn 20 number 2400. preference 0 secondary 9 huntstop call-forward all 4111 call-waiting beep On Mon, Mar 30, 2009 at 10:31 PM, Chris Parker cpar...@cparker.us mailto:cpar...@cparker.us wrote: Duy, Remember, the digit analyzer in IOS will always match the peer with the most matching digits first. So for the first issue of forwarding to VM. The config you have is exactly right. All of the extensions at that site are 4001 - 4003. So when you dial 24001, it will always match that ephone first and go where you want tit to go. The same goes with the WAN dial peer (2300). The extension rages at the other locations are 2001-4 and 3001-4 respectively. The only time you would get into hot water is if you had an extension at HQ that was 2400. Then you would have a problem. For the third requirement the answer is simple make a peer with the destination patten of: 2[23]... Chris Duy Nguyen wrote: How would I achieve this? User at Site C should press 24XXX, then it will forward to user's voice mailbox greeting. E.g. when user dial 24001, then it will be forwarding directly to 4001 VM and leave a message. This call routing should work over the WAN also. My solution: ephone-dn 20 number 24... call-forward all 4111 Problem is: dial-peer voice 2300 voip destination-pattern [23]... session target ras tech-prefix 1 dtmf-relay h245-alphanumeric ip qos dscp cs3 signaling no vad ! Another issue, it is also asking for the same on CCM side when user dials 22XXX should go to VM directly and should also work over the WAN. From CME could not dial 22XXX since it keeps hitting Dial-peer voice 2300 voip. If I change the dial-peer voice 2300 destination-pattern [23][0128].. Still won't achieve calling to Main site phone's VM.
Re: [OSL | CCIE_Voice] gatekeeper question
Hi, I've not tested this since I don't have lab access yet. I can use endpoint max'conn on GK since now I've two trunks towards CCM. But below I described using CCM locations. Both should work depending on what is allowed in GK config snap-shot. gw-priority config would be straight forward: zone-prefix GK 1* gw-priority 10 Sub_Trunk_1 zone-prefix GK 1* gw-priority 9 Pub_Trunk_2 --- On Sun, 3/29/09, CCIE OSL ccie...@gmail.com wrote: From: CCIE OSL ccie...@gmail.com Subject: Re: [OSL | CCIE_Voice] gatekeeper question To: kapil atrish nice_cha...@yahoo.com Cc: ccie Me ccievoic...@yahoo.com, ccie_voice@onlinestudylist.com Date: Sunday, March 29, 2009, 11:43 AM kapil, Actually, I was thinking of using AAR for the first part of the requirement. - As I said before, I have not tried this, I am scheculed for a proctorslab Monday. For the requirement of from HQ to SiteC via GK it will be rejected by GK and rerouted to pstn via 6608 on HQ. I should be able to create AAR group for BR2 and apply it to the trunk. this way I can reserve the 4 digit HQ ANI as well. I may have to use Location but I think GK will send out a call reject to CCM. Have you confirm that your method works. If you got this working, Can you send me GK end and GK gw-prefix output for this? Thanks... /Jin Jung... kapil atrish wrote: You can try this: Create two trunks between CCM and GK having only one CCM in each trunk i.e one with Pub and another one with Sub. Create two set of regions (codec G729), two locations (24kbps to allow only single call over the trunk), and two DPs. Bind all this with respective trunks. On the GK use gw-priority as regular, primary Sub and secondary Pub. For HQ to Site C calling: Create two RGs having Sub and HQGW. Create RL having both these RGs. Create a RP for Site CPoint to this RL. Now if any call is already active over this H.323 Trunk any subsequent call from HQ side will fall back (Location on the GK trunk will reject this call). You might have to turn on the CCM Service parameters (Continue routing on unallocated number). For Site C to HQ Calling: Since 1 call is already active on Sub, any subsequent call from Site C will not be allowed due to b/w limitation over that trunk (Location). Next call should fall back to Pub trunk which is having gw-priority 9. Thanks... --- On *Sun, 3/29/09, CCIE OSL /ccie...@gmail.com/* wrote: From: CCIE OSL ccie...@gmail.com Subject: Re: [OSL | CCIE_Voice] gatekeeper question To: ccie Me ccievoic...@yahoo.com Cc: ccie_voice@onlinestudylist.com Date: Sunday, March 29, 2009, 9:52 AM I have some questions for you. 1. Are you running on 1 tech-prefix or two with on the gatekeeper? 2. Does entire BR2 has to able to call HQ or just a single phone? 3. Does it required to only use 1#, are you allow to use other prefixes? Your first requirement for HQ to BR2 is fairly easy, However, second requirement, is bit confusing. I think in order to make that work, I will have to use Hop-off prefix and statically map a another prefix to PUB address. But since the CAC requirement of single call, and If I were to use bandwidth interzone, I almost need another zone just for PUB, Which means I may have to use different CAC method. /? or somehow allow calls to PUB work using hop-off prefix not affected by GK CAC.??? I have proctor lab coming up on Monday night, I may have to lab this up. If you can provide answer to my questions, It may help me to get this done. Thanks... /Jin Jung... ccie Me wrote: Gents, i'm working on this case on gatekeeper: i need to only allow ONE active call that should be going through SUB now for any other new call if: - it is from HQ to SiteC via GK it will be rejected by GK and rerouted to pstn via 6608 on HQ - if it is from SitC to HQ via GK it will go through PUB instead of SUB i tried to play with regions and CAC on gatekepper and CCM. but i don't think that will lead to solve this case, does any body have idea about this thanks
Re: [OSL | CCIE_Voice] gatekeeper question
You can try this: Create two trunks between CCM and GK having only one CCM in each trunk i.e one with Pub and another one with Sub. Create two set of regions (codec G729), two locations (24kbps to allow only single call over the trunk), and two DPs. Bind all this with respective trunks. On the GK use gw-priority as regular, primary Sub and secondary Pub. For HQ to Site C calling: Create two RGs having Sub and HQGW. Create RL having both these RGs. Create a RP for Site CPoint to this RL. Now if any call is already active over this H.323 Trunk any subsequent call from HQ side will fall back (Location on the GK trunk will reject this call). You might have to turn on the CCM Service parameters (Continue routing on unallocated number). For Site C to HQ Calling: Since 1 call is already active on Sub, any subsequent call from Site C will not be allowed due to b/w limitation over that trunk (Location). Next call should fall back to Pub trunk which is having gw-priority 9. Thanks... --- On Sun, 3/29/09, CCIE OSL ccie...@gmail.com wrote: From: CCIE OSL ccie...@gmail.com Subject: Re: [OSL | CCIE_Voice] gatekeeper question To: ccie Me ccievoic...@yahoo.com Cc: ccie_voice@onlinestudylist.com Date: Sunday, March 29, 2009, 9:52 AM I have some questions for you. 1. Are you running on 1 tech-prefix or two with on the gatekeeper? 2. Does entire BR2 has to able to call HQ or just a single phone? 3. Does it required to only use 1#, are you allow to use other prefixes? Your first requirement for HQ to BR2 is fairly easy, However, second requirement, is bit confusing. I think in order to make that work, I will have to use Hop-off prefix and statically map a another prefix to PUB address. But since the CAC requirement of single call, and If I were to use bandwidth interzone, I almost need another zone just for PUB, Which means I may have to use different CAC method. /? or somehow allow calls to PUB work using hop-off prefix not affected by GK CAC.??? I have proctor lab coming up on Monday night, I may have to lab this up. If you can provide answer to my questions, It may help me to get this done. Thanks... /Jin Jung... ccie Me wrote: Gents, i'm working on this case on gatekeeper: i need to only allow ONE active call that should be going through SUB now for any other new call if: - it is from HQ to SiteC via GK it will be rejected by GK and rerouted to pstn via 6608 on HQ - if it is from SitC to HQ via GK it will go through PUB instead of SUB i tried to play with regions and CAC on gatekepper and CCM. but i don't think that will lead to solve this case, does any body have idea about this thanks
Re: [OSL | CCIE_Voice] gatekeeper question
You are absolutely righttwo ccm groups required having one ccm in each. --- On Sun, 3/29/09, anil batra anil...@yahoo.com wrote: From: anil batra anil...@yahoo.com Subject: Re: [OSL | CCIE_Voice] gatekeeper question To: ccie Me ccievoic...@yahoo.com, CCIE OSL ccie...@gmail.com, kapil atrish nice_cha...@yahoo.com Cc: ccie_voice@onlinestudylist.com Date: Sunday, March 29, 2009, 11:06 AM Do you mean we will be creating two CCM groups, two DP and then assign DP -Pub to one GK trunk and DP-Sub to another GK trunk...if you assgign a DP with CCM group having two CCM's it will be registering the trunk - gk-tunk_1 and gk-trunk_2 bu defualt ... --- On Sun, 3/29/09, kapil atrish nice_cha...@yahoo.com wrote: From: kapil atrish nice_cha...@yahoo.com Subject: Re: [OSL | CCIE_Voice] gatekeeper question To: ccie Me ccievoic...@yahoo.com, CCIE OSL ccie...@gmail.com Cc: ccie_voice@onlinestudylist.com Date: Sunday, March 29, 2009, 10:55 AM You can try this: Create two trunks between CCM and GK having only one CCM in each trunk i.e one with Pub and another one with Sub. Create two set of regions (codec G729), two locations (24kbps to allow only single call over the trunk), and two DPs. Bind all this with respective trunks. On the GK use gw-priority as regular, primary Sub and secondary Pub. For HQ to Site C calling: Create two RGs having Sub and HQGW. Create RL having both these RGs. Create a RP for Site CPoint to this RL. Now if any call is already active over this H.323 Trunk any subsequent call from HQ side will fall back (Location on the GK trunk will reject this call). You might have to turn on the CCM Service parameters (Continue routing on unallocated number). For Site C to HQ Calling: Since 1 call is already active on Sub, any subsequent call from Site C will not be allowed due to b/w limitation over that trunk (Location). Next call should fall back to Pub trunk which is having gw-priority 9. Thanks... --- On Sun, 3/29/09, CCIE OSL ccie...@gmail.com wrote: From: CCIE OSL ccie...@gmail.com Subject: Re: [OSL | CCIE_Voice] gatekeeper question To: ccie Me ccievoic...@yahoo.com Cc: ccie_voice@onlinestudylist.com Date: Sunday, March 29, 2009, 9:52 AM I have some questions for you. 1. Are you running on 1 tech-prefix or two with on the gatekeeper? 2. Does entire BR2 has to able to call HQ or just a single phone? 3. Does it required to only use 1#, are you allow to use other prefixes? Your first requirement for HQ to BR2 is fairly easy, However, second requirement, is bit confusing. I think in order to make that work, I will have to use Hop-off prefix and statically map a another prefix to PUB address. But since the CAC requirement of single call, and If I were to use bandwidth interzone, I almost need another zone just for PUB, Which means I may have to use different CAC method. /? or somehow allow calls to PUB work using hop-off prefix not affected by GK CAC.??? I have proctor lab coming up on Monday night, I may have to lab this up. If you can provide answer to my questions, It may help me to get this done. Thanks... /Jin Jung... ccie Me wrote: Gents, i'm working on this case on gatekeeper: i need to only allow ONE active call that should be going through SUB now for any other new call if: - it is from HQ to SiteC via GK it will be rejected by GK and rerouted to pstn via 6608 on HQ - if it is from SitC to HQ via GK it will go through PUB instead of SUB i tried to play with regions and CAC on gatekepper and CCM. but i don't think that will lead to solve this case, does any body have idea about this thanks
[OSL | CCIE_Voice] Unity - Wait while I transfer option not available
Hi List, Inside Unity Call Handler or SubscriberCall Transfer options you have the Checkbox to enable/disable the prompt Wait while I transfer your call. I noted in a unity system this option is not there at all. Has anybody seen this? Does anyone know if there is a way in Unity to get the option appear or how to make it disappear from Unity GUI interface?
Re: [OSL | CCIE_Voice] Unity - Wait while I transfer option not available
OK. Strangely I've see on all unity servers in PL labs the check box is available however Cisco documentation says its not available before Unity 4.2. http://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_qanda_item09186a0080093c13.shtml I've found a workaround to disable it by replacing the prompt with a blank one. http://forum.cisco.com/eforum/servlet/NetProf?page=netprofforum=Unified%20Communications%20and%20Videotopic=Unified%20Communications%20ApplicationstopicID=.ee835d2fromOutline=CommCmd=MB%3Fcmd%3Ddisplay_location%26location%3D.2cbee5b8 --- On Tue, 3/24/09, Cristobal Priego cristobalpri...@gmail.com wrote: From: Cristobal Priego cristobalpri...@gmail.com Subject: Re: [OSL | CCIE_Voice] Unity - Wait while I transfer option not available To: kapil atrish nice_cha...@yahoo.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Tuesday, March 24, 2009, 7:32 PM What is the unity version? On Mar 24, 2009, at 6:59 AM, kapil atrish nice_cha...@yahoo.com wrote: Hi List, Inside Unity Call Handler or SubscriberCall Transfer options you have the Checkbox to enable/disable the prompt Wait while I transfer your call. I noted in a unity system this option is not there at all. Has anybody seen this? Does anyone know if there is a way in Unity to get the option appear or how to make it disappear from Unity GUI interface?
Re: [OSL | CCIE_Voice] 6500 SERVER ACL
following statement should also be added for H245 traffic: set qos acl ip POD15_SERVER dscp 24 tcp any any ran 11000 65535 --- On Sun, 3/22/09, Christian Hennrich christian.hennr...@intact-is.com wrote: From: Christian Hennrich christian.hennr...@intact-is.com Subject: Re: [OSL | CCIE_Voice] 6500 SERVER ACL To: Chris Parker cpar...@cparker.us Cc: OSL Group ccie_voice@onlinestudylist.com Date: Sunday, March 22, 2009, 2:13 AM Hi Chris, I would mark the RTP also with the ACL: set qos acl ip POD15_SERVER dscp 46 udp any range 16384 32767 any set qos acl ip POD15_SERVER dscp 46 udp any any range 16384 32767 I think, you will not see a question, where you should mark and trust at the same time. And as far as I have tested, does marking only work, when I had set the ports to untrusted. HTH Chris Parker schrieb: Here is my basic 6500 ACL for marking signaling to CS3: set qos acl ip POD15_SERVER dscp 24 tcp any eq 2000 any set qos acl ip POD15_SERVER dscp 24 udp any eq 2427 any set qos acl ip POD15_SERVER dscp 24 tcp any eq 2428 any set qos acl ip POD15_SERVER dscp 24 udp any any eq 5060 set qos acl ip POD15_SERVER dscp 24 tcp any any eq 5060 set qos acl ip POD15_SERVER dscp 24 udp any ran 1718 1720 any set qos acl ip POD15_SERVER dscp 24 tcp any ran 1718 1720 any set qos acl ip POD15_SERVER dscp 24 udp any any ran 1718 1720 set qos acl ip POD15_SERVER dscp 24 tcp any any ran 1718 1720 Should I go ahead and add set qos acl ip POD15_SERVER trust-dscp ip any any to the end of the ACL so that we trust the DSCP marking of any media (MOH, announcements, MTP) that originates from the UCM / Unity servers? Chris __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __
Re: [OSL | CCIE_Voice] Policer on Cat6k - Aggregate or Microflow
Hi list, Has anyone any thoughts on this? --- On Fri, 3/13/09, kapil atrish nice_cha...@yahoo.com wrote: From: kapil atrish nice_cha...@yahoo.com Subject: Policer on Cat6k - Aggregate or Microflow To: ccie_voice@onlinestudylist.com Date: Friday, March 13, 2009, 1:46 AM HI, I am looking for a clarification on the policer to be used on Cat 6k. Question says limit sccp traffic from phones to 32k, I've see few posts where an aggregate policer has been configured and implemented on voice vlan. Shouldn't there be a microflow policer since I want to limit sccp traffic to 32k per phone and not in total? And if it is assumed that we need to limit cumulative sccp traffic to 32k, I've seen when same section asks to limit sccp traffic at BR1 and BR2 also, policing has been implemented inside policy-map which is an individual policer itself. There should've been a aggregate policer created and called inside policy-map??
Re: [OSL | CCIE_Voice] how to make CME phone to have same number on button 1 and 2?
You are not ref to the label, correct? I know of following two ways: 1) You can simply apply the same ephone-dn to button 1 and 2 on same phone. 2) You can create multiple DNs with same number and apply to two different buttons of same ephone. --- On Fri, 3/13/09, CCIE OSL ccie...@gmail.com wrote: From: CCIE OSL ccie...@gmail.com Subject: [OSL | CCIE_Voice] how to make CME phone to have same number on button 1 and 2? To: OSL Group ccie_voice@onlinestudylist.com Date: Friday, March 13, 2009, 12:10 AM how to make CME phone to have same number on button 1 and 2? Is there a way to make 3001 appear on both button 1 and 2? I am not talking about description field. So it will show up 6727653001 --description 3001 -- button 1 3001 -- button 2 /Jin Jung...
Re: [OSL | CCIE_Voice] How to only allow one international or LD call?
Put max-conn under dial-peer. --- On Fri, 3/13/09, CCIE OSL ccie...@gmail.com wrote: From: CCIE OSL ccie...@gmail.com Subject: [OSL | CCIE_Voice] How to only allow one international or LD call? To: OSL Group ccie_voice@onlinestudylist.com Date: Friday, March 13, 2009, 12:56 AM Question How do you only allow one international or LD all any givin time? This is CAC question or COR,, or both??
[OSL | CCIE_Voice] Policer on Cat6k - Aggregate or Microflow
HI, I am looking for a clarification on the policer to be used on Cat 6k. Question says limit sccp traffic from phones to 32k, I've see few posts where an aggregate policer has been configured and implemented on voice vlan. Shouldn't there be a microflow policer since I want to limit sccp traffic to 32k per phone and not in total? And if it is assumed that we need to limit cumulative sccp traffic to 32k, I've seen when same section asks to limit sccp traffic at BR1 and BR2 also, policing has been implemented inside policy-map which is an individual policer itself. There should've been a aggregate policer created and called inside policy-map??
[OSL | CCIE_Voice] CNAME with PSTN
Hi, I've a query regarding CNAME display when calling PSTN. 1) CCM Phone calls PSTN Phone Pstn phone can see the CNAME of CCM Phone but CCM Phone can't see the name of called party. 2) PSTN Phone calls CCM Phone CCM phone can see the CNAME of PSTN phone but PSTN Phone can't see the name of called party. I've display-IE delivery check-box checked on CCM and isdn outgoing display-ie on PSTN side. CLID gets displayed correctly but not the CNAME, is this the correct behaviour? _ Wish to Marry Now? Join MSN Matrimony FREE! http://www.in.msn.com/matrimony
Re: [OSL | CCIE_Voice] DSCP marking on NM-ESW module
I tried match protocol but no good. See below error: Pod28-BR1-RTR(config-if)#service-policy input EF %Error: FastEthernet1/0 Service Policy Configuration Failed.Only Match with access group is supported Date: Mon, 9 Mar 2009 16:46:00 -0800 Subject: Re: [OSL | CCIE_Voice] DSCP marking on NM-ESW module From: vma...@ipexpert.com To: kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com Re: [OSL | CCIE_Voice] DSCP marking on NM-ESW module Can you use “match protocol skinny” and “match protocol rtp”? -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Kapil Atrish kapilatr...@hotmail.com Date: Tue, 10 Mar 2009 02:23:28 +0530 To: OSL Group ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] DSCP marking on NM-ESW module Any other workaround to mark all SCCP and RTP packets to respective DSCP values on this module? I could successfully do it on Cat 3550 with access-list + port range though. _ Windows Live Messenger. Multitasking at its finest. http://www.microsoft.com/india/windows/windowslive/messenger.aspx
Re: [OSL | CCIE_Voice] ATA186 - not able to change anything
OK. I faced this again today, but luckily cisco worked. I put in old password as cisco and new password fields as blank and was able to access the ATA. From: ghaus...@cox.net To: kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] ATA186 - not able to change anything Date: Mon, 9 Mar 2009 23:52:06 -0700 I had this exact problem happen to me on Sunday. There was no way to get around this error. I thought it was just my pod 8… From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kapil Atrish Sent: Saturday, March 07, 2009 2:38 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] ATA186 - not able to change anything Hi list, I've an ATA on POD11 which has port 1 enabled by default. I want to enable Port 2 as well. Whenever I click on SCCP parameters it opens the Change UIP password page. I tried other options under Change Parameters but it still asks for UIP password change. It seems ATA web-administration has been disabled. Does anybody how to overcome this? I tried passwords cisco, 12345 with no luck. thanks Twice the fun— Share photos while you chat with Windows Live Messenger. _ The new Windows Live Messenger. You don’t want to miss this. http://www.microsoft.com/india/windows/windowslive/messenger.aspx
Re: [OSL | CCIE_Voice] Any Solution for: G729 call failed when CCM - CME BACD via Gatekeeper??
You need Xcoder. BACD supports only G711 and since you are doing G729, you'll
face this problem. To verify, make end to end G711 and you should be able to
reach BACD successfully.
--- On Wed, 3/11/09, Jiahong - tobeccie Fang mo...@hotmail.com wrote:
From: Jiahong - tobeccie Fang mo...@hotmail.com
Subject: [OSL | CCIE_Voice] Any Solution for: G729 call failed when CCM - CME
BACD via Gatekeeper??
To: ccie_voice@onlinestudylist.com
Date: Wednesday, March 11, 2009, 12:09 AM
#yiv237988227 .hmmessage P
{
margin:0px;padding:0px;}
#yiv237988227 {
font-size:10pt;font-family:Verdana;}
CCM and CME all register to GK, call between CCM and CME is G729. Call ip
phones between each site work well.
Only problem is: when call from CCM to CME BACD say: DN:8800, it failed. When
debug script in CME, can see 'aa' script did play prompt.
VoIP_CME#
Jan 29 22:09:33.670: //214//TCL :/tcl_PutsObjCmd:
proc init_perCallvars
Jan 29 22:09:33.670:
Jan 29 22:09:33.674: //214//TCL :/tcl_PutsObjCmd: TCL AA: ++ Playing Welcome
Prompt and options menu ++
For gk trunk in CCM site, turn on or off 'wait for far-end h245 terminial
capability set' - same result.
I know this is known issue.
Does anyone have quick solution for it?
Thanks
James
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[OSL | CCIE_Voice] DSCP marking on NM-ESW module
Hi List, I tried to mark SCCP/RTP traffic on ESW module, but the policy got rejected with a message the it doesn't support range keyword in access-list. Any other workaround to mark all SCCP and RTP packets to respective DSCP values on this module? I could successfully do it on Cat 3550 with access-list + port range though. Thanks _ The new Windows Live Messenger. You don’t want to miss this. http://www.microsoft.com/india/windows/windowslive/messenger.aspx
[OSL | CCIE_Voice] ATA registration in SRST Mode
Hi, I observed both ports on ATA register with dual-line even though max-dn dual-line is not configured. Is this expected behavior? I couldn't find much information on cisco reg this. See following capture interface FastEthernet0/0 /call-mana filtering... call-manager-fallback max-conferences 4 gain -6 ip source-address 142.103.65.254 port 2000 strict-match max-ephones 6 max-dn 12 voicemail 14082032600 moh music-on-hold.au multicast moh 239.1.1.1 port 16384 route 142.103.65.254 142.33.65.1 cor incoming int 2 3002 ! BR1(config)#do sh ephone ephone-1 Mac:0013.1978.5969 TCP socket:[1] activeLine:0 REGISTERED in SCCP ver 15 and Server in ver 5 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 IP:142.103.65.54 53094 7970 keepalive 130 max_line 8 button 1: dn 1 number 3002 CM Fallback CH1 IDLE ephone-2 Mac:0011.BB1B.C797 TCP socket:[2] activeLine:1 REGISTERED in SCCP ver 15 and Server in ver 5 mediaActive:0 offhook:1 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 IP:142.103.65.53 52572 7970 keepalive 129 max_line 8 button 1: dn 2 number 3001 CM Fallback CH1 ALERTING Active Call on DN 2 chan 1 :3001 0.0.0.0 0 to 0.0.0.0 2000 via 142.103.65.53 G711Ulaw64k 160 bytes no vad Tx Pkts 0 bytes 0 Rx Pkts 0 bytes 0 Lost 0 Jitter 0 Latency 0 callingDn -1 calledDn 3 ephone-3 Mac:0019.E7B7.C637 TCP socket:[3] activeLine:0 REGISTERED in SCCP ver 0 and Server in ver 0 mediaActive:0 offhook:0 ringing:1 reset:0 reset_sent:0 paging 0 debug:0 IP:142.103.65.55 7642 ATA Phone keepalive 121 max_line 2 dual-line button 1: dn 3 number 3006 CM Fallback CH1 RINGING button 2: dn 4 number 3006 CM Fallback CH1 IDLE call ringing on line 1 ephone-4 Mac:19E7.B7C6.3701 TCP socket:[4] activeLine:0 REGISTERED in SCCP ver 0 and Server in ver 0 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 IP:142.103.65.55 6924 ATA Phone keepalive 111 max_line 2 dual-line button 1: dn 5 number 3007 CM Fallback CH1 IDLE button 2: dn 6 number 3007 CM Fallback CH1 IDLE BR1(config)#do sh ephone ephone-1 Mac:0013.1978.5969 TCP socket:[1] activeLine:0 REGISTERED in SCCP ver 15 and Server in ver 5 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 IP:142.103.65.54 53094 7970 keepalive 130 max_line 8 button 1: dn 1 number 3002 CM Fallback CH1 IDLE ephone-2 Mac:0011.BB1B.C797 TCP socket:[2] activeLine:1 REGISTERED in SCCP ver 15 and Server in ver 5 mediaActive:1 offhook:1 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 IP:142.103.65.53 52572 7970 keepalive 130 max_line 8 button 1: dn 2 number 3001 CM Fallback CH1 CONNECTED Active Call on DN 2 chan 1 :3001 142.103.65.53 20468 to 142.103.65.55 16385 via 142.103.65.53 G711Ulaw64k 160 bytes no vad Tx Pkts 494 bytes 84968 Rx Pkts 492 bytes 84624 Lost 0 Jitter 0 Latency 0 callingDn -1 calledDn 3 ephone-3 Mac:0019.E7B7.C637 TCP socket:[3] activeLine:1 REGISTERED in SCCP ver 0 and Server in ver 0 mediaActive:1 offhook:1 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 IP:142.103.65.55 7642 ATA Phone keepalive 121 max_line 2 dual-line button 1: dn 3 number 3006 CM Fallback CH1 CONNECTED button 2: dn 4 number 3006 CM Fallback CH1 IDLE Active Call on DN 3 chan 1 :3006 142.103.65.55 16385 to 142.103.65.53 20468 via 142.103.65.55 G711Ulaw64k 160 bytes no vad Tx Pkts 0 bytes 0 Rx Pkts 0 bytes 0 Lost 0 Jitter 0 Latency 0 callingDn 2 calledDn -1 ephone-4 Mac:19E7.B7C6.3701 TCP socket:[4] activeLine:0 REGISTERED in SCCP ver 0 and Server in ver 0 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 IP:142.103.65.55 6924 ATA Phone keepalive 111 max_line 2 dual-line button 1: dn 5 number 3007 CM Fallback CH1 IDLE button 2: dn 6 number 3007 CM Fallback CH1 IDLE BR1(config)#do sh ephone ephone-1 Mac:0013.1978.5969 TCP socket:[1] activeLine:0 REGISTERED in SCCP ver 15 and Server in ver 5 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 IP:142.103.65.54 53094 7970 keepalive 131 max_line 8 button 1: dn 1 number 3002 CM Fallback CH1 IDLE ephone-2 Mac:0011.BB1B.C797 TCP socket:[2] activeLine:0 REGISTERED in SCCP ver 15 and Server in ver 5 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 IP:142.103.65.53 52572 7970 keepalive 131 max_line 8 button 1: dn 2 number 3001 CM Fallback CH1 IDLE ephone-3 Mac:0019.E7B7.C637 TCP socket:[3] activeLine:0 REGISTERED in SCCP ver 0 and Server in ver 0 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 IP:142.103.65.55 7642 ATA Phone keepalive 122 max_line 2 dual-line button 1: dn 3 number 3006 CM Fallback CH1 IDLE button 2: dn 4 number 3006 CM Fallback CH1 IDLE ephone-4 Mac:19E7.B7C6.3701 TCP socket:[4] activeLine:0 REGISTERED in SCCP ver 0 and Server in ver 0 mediaActive:0 offhook:0
Re: [OSL | CCIE_Voice] ATA registration in SRST Mode
After switching over to max-dn dual-line I realised ATA actually registers with two channels now. Earlier it showed two buttons both associated to Channel 1. If someone can comment on the usage of two buttons with single channe or if there is something wrongl, that'll be great. See below output when max-dn with dual-line configured /call-mana filtering... call-manager-fallback max-conferences 4 gain -6 ip source-address 142.103.65.254 port 2000 strict-match max-ephones 6 max-dn 15 dual-line voicemail 14082032600 moh music-on-hold.au multicast moh 239.1.1.1 port 16384 route 142.103.65.254 142.33.65.1 cor incoming int 2 3002 ! BR1(config-cm-fallback)#do sh ephone ephone-1 Mac:0013.1978.5969 TCP socket:[1] activeLine:0 REGISTERED in SCCP ver 15 and Server in ver 5 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 IP:142.103.65.54 51294 7970 keepalive 38 max_line 8 button 1: dn 1 number 3002 CM Fallback CH1 IDLE CH2 IDLE ephone-2 Mac:0011.BB1B.C797 TCP socket:[2] activeLine:0 REGISTERED in SCCP ver 15 and Server in ver 5 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 IP:142.103.65.53 52527 7970 keepalive 39 max_line 8 button 1: dn 2 number 3001 CM Fallback CH1 IDLE CH2 IDLE ephone-3 Mac:0019.E7B7.C637 TCP socket:[3] activeLine:0 REGISTERED in SCCP ver 3 and Server in ver 3 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 IP:142.103.65.55 7003 ATA Phone keepalive 23 max_line 1 button 1: dn 3 number 3006 CM Fallback CH1 IDLE CH2 IDLE ephone-4 Mac:19E7.B7C6.3701 TCP socket:[4] activeLine:0 REGISTERED in SCCP ver 3 and Server in ver 3 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 IP:142.103.65.55 6379 ATA Phone keepalive 13 max_line 1 button 1: dn 4 number 3007 CM Fallback CH1 IDLE CH2 IDLE Thanks... From: kapilatr...@hotmail.com To: ccie_voice@onlinestudylist.com Subject: ATA registration in SRST Mode Date: Mon, 2 Mar 2009 14:49:24 +0530 Hi, I observed both ports on ATA register with dual-line even though max-dn dual-line is not configured. Is this expected behavior? I couldn't find much information on cisco reg this. See following capture interface FastEthernet0/0 /call-mana filtering... call-manager-fallback max-conferences 4 gain -6 ip source-address 142.103.65.254 port 2000 strict-match max-ephones 6 max-dn 12 voicemail 14082032600 moh music-on-hold.au multicast moh 239.1.1.1 port 16384 route 142.103.65.254 142.33.65.1 cor incoming int 2 3002 ! BR1(config)#do sh ephone ephone-1 Mac:0013.1978.5969 TCP socket:[1] activeLine:0 REGISTERED in SCCP ver 15 and Server in ver 5 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 IP:142.103.65.54 53094 7970 keepalive 130 max_line 8 button 1: dn 1 number 3002 CM Fallback CH1 IDLE ephone-2 Mac:0011.BB1B.C797 TCP socket:[2] activeLine:1 REGISTERED in SCCP ver 15 and Server in ver 5 mediaActive:0 offhook:1 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 IP:142.103.65.53 52572 7970 keepalive 129 max_line 8 button 1: dn 2 number 3001 CM Fallback CH1 ALERTING Active Call on DN 2 chan 1 :3001 0.0.0.0 0 to 0.0.0.0 2000 via 142.103.65.53 G711Ulaw64k 160 bytes no vad Tx Pkts 0 bytes 0 Rx Pkts 0 bytes 0 Lost 0 Jitter 0 Latency 0 callingDn -1 calledDn 3 ephone-3 Mac:0019.E7B7.C637 TCP socket:[3] activeLine:0 REGISTERED in SCCP ver 0 and Server in ver 0 mediaActive:0 offhook:0 ringing:1 reset:0 reset_sent:0 paging 0 debug:0 IP:142.103.65.55 7642 ATA Phone keepalive 121 max_line 2 dual-line button 1: dn 3 number 3006 CM Fallback CH1 RINGING button 2: dn 4 number 3006 CM Fallback CH1 IDLE call ringing on line 1 ephone-4 Mac:19E7.B7C6.3701 TCP socket:[4] activeLine:0 REGISTERED in SCCP ver 0 and Server in ver 0 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 IP:142.103.65.55 6924 ATA Phone keepalive 111 max_line 2 dual-line button 1: dn 5 number 3007 CM Fallback CH1 IDLE button 2: dn 6 number 3007 CM Fallback CH1 IDLE BR1(config)#do sh ephone ephone-1 Mac:0013.1978.5969 TCP socket:[1] activeLine:0 REGISTERED in SCCP ver 15 and Server in ver 5 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 IP:142.103.65.54 53094 7970 keepalive 130 max_line 8 button 1: dn 1 number 3002 CM Fallback CH1 IDLE ephone-2 Mac:0011.BB1B.C797 TCP socket:[2] activeLine:1 REGISTERED in SCCP ver 15 and Server in ver 5 mediaActive:1 offhook:1 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 IP:142.103.65.53 52572 7970 keepalive 130 max_line 8 button 1: dn 2 number 3001 CM Fallback CH1 CONNECTED Active Call on DN 2 chan 1 :3001 142.103.65.53 20468 to 142.103.65.55 16385 via 142.103.65.53 G711Ulaw64k 160 bytes no vad Tx Pkts 494 bytes 84968 Rx Pkts 492 bytes
Re: [OSL | CCIE_Voice] 6608 gw
In the route pattern or route-group level, don't use Use external phone number
mask and put area-code (3 digits) as Prefix to calling number.
--- On Mon, 3/2/09, hasan khalife hasan_khal...@hotmail.com wrote:
From: hasan khalife hasan_khal...@hotmail.com
Subject: [OSL | CCIE_Voice] 6608 gw
To: ccie_voice@onlinestudylist.com
Date: Monday, March 2, 2009, 10:55 PM
#yiv1601154501 .hmmessage P
{
margin:0px;padding:0px;}
#yiv1601154501 {
font-size:10pt;font-family:Verdana;}
1-when calling name should not be displaY juSt unckeck the IE BOX ?
2- 7 DIGIT CALLING NUMBER SHOULD BE DISPLAY ON THE PSTN ?where is must
specify ?
thx
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Re: [OSL | CCIE_Voice] Incoming TEHO for HQ from BR2 (Spain) not as advertised.
Instead of rolling it over to HQ Unrestricted, I created a CSS having access to only 1x which internally points to HQ GW. The TP can only be accessed from GK and am doing PreDot 1# at TP level. It always works for me. Another TP which looks for 1#.[1-2]xxx. I do a PreDot and roll it over to CSS internal. --- On Tue, 3/3/09, Cliff McGlamry cl...@mcglamry.net wrote: From: Cliff McGlamry cl...@mcglamry.net Subject: [OSL | CCIE_Voice] Incoming TEHO for HQ from BR2 (Spain) not as advertised. To: ccie_voice@onlinestudylist.com Date: Tuesday, March 3, 2009, 12:46 AM I've got TEHO set up and working coming in from my BR2 (Spain) site. The number arrives at CallManager as a string 1#1212224 Based on the proctor guide, I set up a translation pattern for 1#1XX and made sure I created it last. However, it wouldn't work. I had the translation pattern set up as 1#.1XX and set it to strip predot and prefix a 9, and then roll it into my Unrestricted HQ CSS. At that point, it should have been able to see the route patterns set up on my various route lists. Unfortunately, it wouldn't work. I attempted multiple permutations of this, none of them successful. I ended up tightening up the translation pattern for internal numbers to be 1#[12]0XX and then created a route pattern for 1#.1[2-9]X stripping pre dot so that the two did not overlap. While this setup did not work in a translation pattern, it DID work in a route pattern. I'm trying to figure out why the setup as a translation pattern would not work. Has anyone else run into this? I'm less concerned with getting an exact match of the provided solution as I am understanding what would cause my setup of it (attempting to duplicate it) would fail. Mark says that the functionality is what's required, and I was able to do that. But why wouldn't my translation pattern work? I even deleted it and added it back again...and got exactly the same results. Cliff
[OSL | CCIE_Voice] IPIP Gatway or CME transcoding not working
Hi list, Actually, it all started when I tried to send a G.729 stream to CME with G.711 at CCM side. I initially had Xcoder on CME but never got invoked. I tried IPIP GW but same result. Below is my scneario and results of the testing so far: I've a trunk from CCM to GK with codec G.711 set. CME also registers to same GK with dial-peers (inbound/outbound) set for default G.729. I've configured IPIP Gateway on GK with transcoders and trying to transcode the calls locally. I've xcoder registered but don't see them getting invoked. If I make CCM Trunk G.711 CME G.711 all works fine, If I make CCM Trunk G.729 CME G.729 all works fine, If I make CCM Trunk G.711 --- CME G.729, both ways call mature on G.729 If I make CCM Trunk G.729 --- CME G.711, I get fast busy both ways. The call disconnect cause-value=47 on CME. It seems CME is trying to Xcode call locally which should've been done by GK. When I tried Xcoding at CME same result/same cause-code. Attached snapshot of some relevant commands. Any input is highly appreciated. IPIP Gateway ! voice-card 1 dsp services dspfarm ! ip cef ! !! voice service voip allow-connections h323 to h323 ! ! interface FastEthernet0/0.103 encapsulation dot1Q 103 ip address 142.103.64.254 255.255.255.0 ip helper-address 142.3.64.11 ip helper-address 142.3.64.12 h323-gateway voip interface h323-gateway voip id IPIP ipaddr 142.33.64.1 1719 h323-gateway voip h323-id IPIP ! ! sccp local FastEthernet0/0.103 sccp sccp ccm 142.33.64.1 priority 1 sccp codec g729ar8 mask sccp codec g729abr8 mask ! dspfarm transcoder maximum sessions 4 dspfarm connection interval 60 dspfarm ! ! gateway ! ! ! gatekeeper zone local IPIP cisco.com 142.33.64.1 zone local ucm cisco.com zone local CME cisco.com invia IPIP outvia IPIP enable-intrazone zone prefix ucm 2* gw-priority 10 gk-trunk_2 zone prefix ucm 3* gw-priority 10 gk-trunk_2 zone prefix CME 4* gw-priority 10 gw gw-type-prefix 1#* default-technology no shutdown ! ! telephony-service max-ephones 1 max-dn 1 ip source-address 142.33.64.1 port 2000 sdspfarm units 1 sdspfarm transcode sessions 4 sdspfarm tag 1 mtp001200d86500 create cnf-files version-stamp Jan 01 2002 00:00:00 max-conferences 4 gain -6 ! HQ#sh sdspfarm sessions summary max-mtps:1, max-streams:8, alloc-streams:8, act-streams:0 ID MTP State CallID confID Usage Codec/Duration = == === == = == 11 IDLE -1 0 G711Ulaw64k /20ms 21 IDLE -1 0 G711Ulaw64k /20ms 31 IDLE -1 0 G711Ulaw64k /20ms 41 IDLE -1 0 G711Ulaw64k /20ms 51 IDLE -1 0 G711Ulaw64k /20ms 61 IDLE -1 0 G711Ulaw64k /20ms 71 IDLE -1 0 G711Ulaw64k /20ms 81 IDLE -1 0 G711Ulaw64k /20ms HQ# HQ# HQ#show sccp ? all Display all SCCP global info connections Display SCCP connections statistics Display SCCP statistics |Output modifiers cr HQ#show sccp conn HQ#show sccp connections ? | Output modifiers cr HQ#show sccp connections Total number of active session(s) 0, and connection(s) 0 HQ#show sccp al HQ#show sccp all SCCP Admin State: UP Gateway IP Address: 142.103.64.254 Switchover Method: IMMEDIATE, Switchback Method: GUARD_TIMER Switchback Guard Timer: 1200 sec, IP Precedence: 5 Max Supported MTP sessions: 0 User Masked Codec list: g729ar8 g729abr8 Transcoding Oper State: ACTIVE - Cause Code: NONE Active Call Manager: 142.33.64.1, Port Number: 2000 TCP Link Status: CONNECTED Conferencing Oper State: DOWN - Cause Code: DSPFARM_DOWN Active Call Manager: NONE TCP Link Status: NOT_CONNECTED Call Manager: 142.33.64.1, Port Number: 2000 Priority: 1, Version: 3.1 or Higher SCCP Transcoding Application Statistics: TCP packets rx 179, tx 185 Unsupported pkts rx 2, Unrecognized pkts rx 0 Register tx 2, successful 2, rejected 0, failed 0 KeepAlive tx 175, successful 175, failed 0 OpenReceiveChannel rx 0, successful 0, failed 0 CloseReceiveChannel rx 0, successful 0, failed 0 StartMediaTransmission rx 0, successful 0, failed 0 StopMediaTransmission rx 0, successful 0, failed 0 MediaStreamingFailure rx 0 Switchover 0, Switchback 0 SCCP Conferencing Application Statistics: TCP packets rx 117, tx 121 Unsupported pkts rx 0, Unrecognized pkts rx 0 Register tx 1, successful 1, rejected 0, failed 0 KeepAlive tx 116, successful 116, failed 0 OpenReceiveChannel rx 0, successful 0, failed 0 CloseReceiveChannel rx
Re: [OSL | CCIE_Voice] BR1 - SRST???
Put following on BR1 to trigger SRST: ip route ccm pub ip /mask null 0 ip route ccm sub ip/mask null 0 --- On Sun, 3/1/09, Mike Brooks 2xcci...@gmail.com wrote: From: Mike Brooks 2xcci...@gmail.com Subject: Re: [OSL | CCIE_Voice] BR1 - SRST??? To: Cliff McGlamry cl...@mcglamry.net Cc: ccie_voice@onlinestudylist.com Date: Sunday, March 1, 2009, 6:30 AM Hey Cliff, There is a couple ways of doing it, but what I always do is just configure a call-manager group with just the SUB in it and put it int the BR1 device pool. Then assign all devices in the BR1 site to the BR1_DP (IP Blue as well). Also, only configure the BR1 router to communicate with the SUB. Then when you are ready to test SRST mode then just stop the CM Service on the SUB. Of course, in the real lab you would just shut down the WAN link and test. hth, Mike Brooks CCIE#16027 (RS) On Sun, Mar 1, 2009 at 8:52 AM, Cliff McGlamry cl...@mcglamry.net wrote: How do you throw BR1 into SRST so that IP Blue will register to the BR1 router but still be able to go via PSTN to Unity?
Re: [OSL | CCIE_Voice] CCM to GK trunkcan load balancing be achieved
OK if question pre-specifies H.323 IDs to be ccm_1 and ccm_2? Since both the trunks can't have same name is this scenario valid and has any workaround. Date: Fri, 20 Feb 2009 11:04:02 -0800 Subject: Re: [OSL | CCIE_Voice] CCM to GK trunkcan load balancing be achieved From: vma...@ipexpert.com To: anil...@yahoo.com; ccie_voice@onlinestudylist.com; kapilatr...@hotmail.com Re: [OSL | CCIE_Voice] CCM to GK trunkcan load balancing be achieved Two trunks: pub_trunk in DP containing CCM Group = PUB-ONLY Sub_trunk in DP containing CCM Group = SUB-ONLY RP RL RG 2 x trunks with circular hunting. -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: anil batra anil...@yahoo.com Reply-To: anil...@yahoo.com Date: Fri, 20 Feb 2009 10:08:12 -0800 (PST) To: OSL Group ccie_voice@onlinestudylist.com, Kapil Atrish kapilatr...@hotmail.com Subject: Re: [OSL | CCIE_Voice] CCM to GK trunkcan load balancing be achieved Not sure if it possible, In CCM when we create RG-GK we can't define HQ_Trunk_1 or HQ_Trunk_2 it is simply HQ-Trunk so the call will go thrugh the same trunk and since the trunk has DP which has CCM group which has SUB is primary so the call will always go thru SUB and only will go thru PUB when SUB fails. --- On Fri, 2/20/09, Kapil Atrish kapilatr...@hotmail.com wrote: From: Kapil Atrish kapilatr...@hotmail.com Subject: [OSL | CCIE_Voice] CCM to GK trunkcan load balancing be achieved To: ccie_voice@onlinestudylist.com Date: Friday, February 20, 2009, 11:28 PM Hi, CCM cluster (pub, sub) is registered to a GK. GK correctly shows two trunks with _1 and _2. Requirement is to enable load-balancing of outgoing calls from CCM to GK over those two trunks , how can it be achieved? I can achieve GK to CCM load-balancing via gw-priority, but requirement is to achieve load blanacing of calls the other way. Sub is my primary call processing agent. Any inputs are highly appreciated... Get a view of the world through MSN Video. Some things just cannot be left unseen. Try it! http://video.msn.com/?mkt=en-in _ For the freshest Indian Jobs Visit MSN Jobs http://www.in.msn.com/jobs
[OSL | CCIE_Voice] MOH from BR1 Flash in case Primary CCM down
Hi List, I want to confirm if there is a way to play MOH from BR1 flash in case Primary MOH server is down. thanks, _ Chose your Life Partner! Join MSN Matrimony FREE http://www.in.msn.com/matrimony
Re: [OSL | CCIE_Voice] MOH from BR1 Flash in case Primary CCM down
Thanks man for your quick reply. Actually BR1 is not in SRST mode. I am playing multicast MOH from Router's flash. Let's say I am using Multicast IP/base port of Sub and sub goes down, is it possible for BR1 to continue playing multicast MOH from its flash. I couldn't find a wayout to make it work. Date: Wed, 25 Feb 2009 13:20:35 +0100 Subject: Re: [OSL | CCIE_Voice] MOH from BR1 Flash in case Primary CCM down From: basant.ya...@gmail.com To: kapilatr...@hotmail.com CC: ccie_voice@onlinestudylist.com Hi Kapil Here is a link that will help you:- http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_feature_guide09186a00802d1c31.html#wp1118060 Regards - Basant On Wed, Feb 25, 2009 at 12:44 PM, Kapil Atrish kapilatr...@hotmail.com wrote: Hi List, I want to confirm if there is a way to play MOH from BR1 flash in case Primary MOH server is down. thanks, Akshay Kumar takes on the two reigning Bollywood Khans. Catch the action on MSN Entertainment! Check it out! _ For the freshest Indian Jobs Visit MSN Jobs http://www.in.msn.com/jobs
Re: [OSL | CCIE_Voice] MOH from BR1 Flash in case Primary CCM down
Yeah but once Sub is offline, the Multicast Address configured at BR1 is not active. How do I point it to Pub dynamically. I tried giving same Address to Pub/Sub and changing Base port 16384 and 16386 but haven't worked since BR1 is still configured for Sub Multicast IP and port no. Is there a wayout I can add two multicast entries in BR1? From: narinder.ku...@uxcg.com.au To: narinder.ku...@uxcg.com.au; kapilatr...@hotmail.com; basant.ya...@gmail.com CC: ccie_voice@onlinestudylist.com Date: Thu, 26 Feb 2009 00:06:45 +1100 Subject: RE: [OSL | CCIE_Voice] MOH from BR1 Flash in case Primary CCM down Kapil correction. No need of route filers, You can have PUB Hop count to 1 in that case 239.1.1.1 will never reach Branch. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kumar, Narinder Sent: Thursday, 26 February 2009 12:04 AM To: Kapil Atrish; basant.ya...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MOH from BR1 Flash in case Primary CCM down Kapil, I haven’t tested this but . If Sub unicast to BR1 than you can easily achieve this by spoofing the PUB multicast add and have PUB ur second choice for MOH for Branch and have multicast hop of 1, and config multicast MOH under call-manager-fall Assuming you are trying to achieve Multicast MOH end to end. 2 MOH Pub and Sub, in 2 different MGR both multicast enabled. Say Sub is ur primary MOH with Base IP add 239.1.2.1 Pub with base IP add 239.1.1.1 Now you need multicast end to end In the MRGL Sub MOH first and PUB MOH second. Now in normal situation Branch PH’s are getting MOH from SUB. In case Sub stop working. You can add some route filers to stop reaching the Pub base address 239.1.1.1 to the branch . The CCM will still tell the phones to listen to 239.1.1.1 and you can spoof it from flash. I don’t know if this will work or not. I haven’t tested myself. But if you have only ONE MOH and you want Branch router as backup, I am not sure if that can be achieved. Cheers Narinder From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kapil Atrish Sent: Wednesday, 25 February 2009 11:43 PM To: basant.ya...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MOH from BR1 Flash in case Primary CCM down Thanks man for your quick reply. Actually BR1 is not in SRST mode. I am playing multicast MOH from Router's flash. Let's say I am using Multicast IP/base port of Sub and sub goes down, is it possible for BR1 to continue playing multicast MOH from its flash. I couldn't find a wayout to make it work. Date: Wed, 25 Feb 2009 13:20:35 +0100 Subject: Re: [OSL | CCIE_Voice] MOH from BR1 Flash in case Primary CCM down From: basant.ya...@gmail.com To: kapilatr...@hotmail.com CC: ccie_voice@onlinestudylist.com Hi Kapil Here is a link that will help you:- http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_feature_guide09186a00802d1c31.html#wp1118060 Regards - Basant On Wed, Feb 25, 2009 at 12:44 PM, Kapil Atrish kapilatr...@hotmail.com wrote: Hi List, I want to confirm if there is a way to play MOH from BR1 flash in case Primary MOH server is down. thanks, Akshay Kumar takes on the two reigning Bollywood Khans. Catch the action on MSN Entertainment! Check it out! Discover your phone style WIN a Windows Mobile phone. Your style! Try it now! CONFIDENTIALITY - The information contained in this electronic mail message is confidential and is intended solely for the addressee(s). If you are not an authorised recipient of this message please contact Getronics Australia immediately by reply email and destroy/delete this message from your computer. Any unauthorised form of reproduction of this message, or part thereof, is strictly prohibited. DISCLAIMER - Unless specifically indicated otherwise, the views and opinions expressed in this email are those of the sender and not Getronics Australia. While we endeavour to protect our network from computer viruses, Getronics Australia does not warrant that this email or any attachments are free of viruses or any other defects or errors. It is the duty of the recipient to virus scan and otherwise test any information contained in this email before loading onto any computer system. CONFIDENTIALITY - The information contained in this electronic mail message is confidential and is intended solely for the addressee(s). If you are not an authorised recipient of this message please contact Getronics Australia immediately by reply email and destroy/delete this message from your computer. Any unauthorised form of reproduction of this message, or part thereof, is strictly prohibited. DISCLAIMER - Unless specifically indicated otherwise, the views and opinions expressed in this email are those of the sender and not Getronics
Re: [OSL | CCIE_Voice] UnityUTIMMessage Notification VS Dialout MWI
OK. I tested this and its indeed 4 digits of ANI. If I configure 4 digits the loop is broken, and even if I configure e.164 loop is still broken. It seems if Unity knows the Port number before hand, whether 4 digits or 10 digits it is preventing the loop rightly. Date: Thu, 19 Feb 2009 08:34:36 -0800 Subject: Re: [OSL | CCIE_Voice] UnityUTIMMessage Notification VS Dialout MWI From: vma...@ipexpert.com To: kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com Re: [OSL | CCIE_Voice] UnityUTIMMessage Notification VS Dialout MWI I think 4 digit # in the first column within UTIM. The best way to make sure is use Call Viewer on Unity and have Unity originate a call to a phone which is fwded to send the call back to Unity. What is the ANI for the call being returned? -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Kapil Atrish kapilatr...@hotmail.com Date: Thu, 19 Feb 2009 09:59:28 +0530 To: vma...@ipexpert.com, ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] UnityUTIMMessage Notification VS Dialout MWI OK. That means if I un-check Message Notification or let it be checked on voice-mail ports, there is no harm as long as I implement the fix to avoid Unity loop, correct? I've one more quick question, do I need to put full e.164 number or only 4 digit number on voice-mail ports to avoid looping of any external call. I checked CCM traces and it showed me 4 digit number as well as FQDN field which is e.164 10 digit number of the VM ports. I couldn't find a way in Unity to see what VM port number it is receiving? Thanks for your inputs Vik.. Date: Wed, 18 Feb 2009 18:13:26 -0800 Subject: Re: [OSL | CCIE_Voice] UnityUTIMMessage Notification VS Dialout MWI From: vma...@ipexpert.com To: kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com Re: [OSL | CCIE_Voice] UnityUTIMMessage Notification VS Dialout MWI Message Notification is different to MWI as you correctly state. When we talk about Unity Looping it is normally related to loops associated with Msg Notification but it can be any call Unity originates that is returned to Unity. See this link for the fix: http://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_tech_note09186a0080094b01.shtml#p3a -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com http://ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Kapil Atrish kapilatr...@hotmail.com http://hotmail.com Date: Wed, 18 Feb 2009 23:45:52 +0530 To: ccie_voice@onlinestudylist.com http://onlinestudylist.com Subject: [OSL | CCIE_Voice] UnityUTIMMessage Notification VS Dialout MWI Hi List, I'v a question about Message notification, I read about it, pl let me know if my understanding is correct. I understand Message Notification ports are used when susbcriber has notifications enabled on additional devices under SubscriberMessage notification settings, like Home Phone/Pager etc. So if I don't have notification enabled on additional devices do I still need to uncheck Message Notification on Ports to avoid Unity looping or is there something I am missing? MWI has nothing to do with Message Notification ports but only with Dialout MWIports, correct? Thanks in advance... Get the latest buzz on outsourcing. Up to date information on mergers, acquisitions and deals on BPO Watch. Try it now! http://www.bpowatchindia.com/default.asp Watch useful tips on recipes, fitness, yoga and fashion only on MSN videos. Try it! http://video.msn.com/?mkt=en-in _ Wish to Marry Now? Join MSN Matrimony FREE! http://www.in.msn.com/matrimony
[OSL | CCIE_Voice] FRTS using MQC is class-default required
Hi List, Quick question on LLQ with FRTS through MQC. I am not able to recall but I read some where the class-default is not required when FRTS is implemented using MQC alongwith LLQ. Below is the config shap-shot: policy-map voice class EF priority percent 50 class AF bandwidth percent 5 class class-defaultIs this required fair-queue ! policy-map FRTS class class-default shape average xx x shape adaptive shape fr-voice-adapt deactivation 30 service-policy voice ! Appreciate any comment on this _ For the freshest Indian Jobs Visit MSN Jobs http://www.in.msn.com/jobs
Re: [OSL | CCIE_Voice] FRTS using MQC is class-default required
Thanks Vik for the clarification. I am not sure where I read but if I manage to get the source info, I'll def share. Date: Fri, 20 Feb 2009 06:38:01 -0800 Subject: Re: [OSL | CCIE_Voice] FRTS using MQC is class-default required From: vma...@ipexpert.com To: kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com Re: [OSL | CCIE_Voice] FRTS using MQC is class-default required I think it is required if you wanted to subject the default class to flow-based fair queuing. If you can retrieve the source that says you don’t then I’ll appreciate that. -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Kapil Atrish kapilatr...@hotmail.com Date: Fri, 20 Feb 2009 17:08:26 +0530 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] FRTS using MQC is class-default required Hi List, Quick question on LLQ with FRTS through MQC. I am not able to recall but I read some where the class-default is not required when FRTS is implemented using MQC alongwith LLQ. Below is the config shap-shot: policy-map voice class EF priority percent 50 class AF bandwidth percent 5 class class-defaultIs this required fair-queue ! policy-map FRTS class class-default shape average xx x shape adaptive shape fr-voice-adapt deactivation 30 service-policy voice ! Appreciate any comment on this Discover your phone style WIN a Windows Mobile phone. Your style! Try it now! http://www.whatsmyphonestyle.com _ For the freshest Indian Jobs Visit MSN Jobs http://www.in.msn.com/jobs
[OSL | CCIE_Voice] CCM to GK trunkcan load balancing be achieved
Hi, CCM cluster (pub, sub) is registered to a GK. GK correctly shows two trunks with _1 and _2. Requirement is to enable load-balancing of outgoing calls from CCM to GK over those two trunks , how can it be achieved? I can achieve GK to CCM load-balancing via gw-priority, but requirement is to achieve load blanacing of calls the other way. Sub is my primary call processing agent. Any inputs are highly appreciated... _ Find a better job. We have plenty. Visit MSN Jobs http://www.in.msn.com/jobs
[OSL | CCIE_Voice] CUE problem on POD 26
Hi List, My cue module is in rebooting state. Reboot of Router didn't help. It always comes to this stage and halt: System Now Booting ...[BOOT-ASM] 7 Please enter '***' to change boot configuration: __ I've observed following during reboot. Verifying signature now... Signature not a valid base64 encoded entity Invalid Kernel Signature !!! Rebooting Has anybody any idea what's wrong. My config is: interface Service-Engine0/0 ip unnumbered FastEthernet0/0.360 service-module ip address 10.26.202.2 255.255.255.0 service-module ip default-gateway 10.26.202.1 !
[OSL | CCIE_Voice] UnityUTIMMessage Notification VS Dialout MWI
Hi List, I'v a question about Message notification, I read about it, pl let me know if my understanding is correct. I understand Message Notification ports are used when susbcriber has notifications enabled on additional devices under SubscriberMessage notification settings, like Home Phone/Pager etc. So if I don't have notification enabled on additional devices do I still need to uncheck Message Notification on Ports to avoid Unity looping or is there something I am missing? MWI has nothing to do with Message Notification ports but only with Dialout MWIports, correct? Thanks in advance... _ Want to explore the world? Visit MSN Travel for the best deals. http://in.msn.com/coxandkings
Re: [OSL | CCIE_Voice] UnityUTIMMessage Notification VS Dialout MWI
Thanks Greg, I think there are two scenarios, one when you've to prevent looping of external calls which land on Unity and then looped from there onwards. Those calls are not originated by unity initially. To fix this you need to apply the solution you mentioned below. Any idea, does it have to be e.164 or 4 digit number, say I've external phone no mask configured for VM ports? Second scenario, when Unity originates the calls by way of message notification and it gets looped back. You need to apply the fix give by Vik in URL below. Pl correct me if you find missing link. From: gpu...@doc.gov To: kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com Date: Wed, 18 Feb 2009 13:52:28 -0500 Subject: RE: [OSL | CCIE_Voice] UnityUTIMMessage Notification VS Dialout MWI You are correct that MWI has no related to Message Notification. To help prevent Unity from looping through its ports you can goto UTIM and on the ports page, put the DN for the specific port(s) as they are configured in CCM. This will allow Unity to know the call came from one of its ports and not to loop back to it. greg -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kapil Atrish Sent: Wednesday, February 18, 2009 1:16 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] UnityUTIMMessage Notification VS Dialout MWI Hi List, I'v a question about Message notification, I read about it, pl let me know if my understanding is correct. I understand Message Notification ports are used when susbcriber has notifications enabled on additional devices under SubscriberMessage notification settings, like Home Phone/Pager etc. So if I don't have notification enabled on additional devices do I still need to uncheck Message Notification on Ports to avoid Unity looping or is there something I am missing? MWI has nothing to do with Message Notification ports but only with Dialout MWIports, correct? Thanks in advance... Get the latest buzz on outsourcing. Up to date information on mergers, acquisitions and deals on BPO Watch. Try it now! http://www.bpowatchindia.com/default.asp _ Movies, sports news! Get your daily entertainment fix, only on live.com http://www.live.com/?scope=videoform=MICOAL
Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
Yup, you are right. Those are not under English_United States folder. You'll find ANN_Fastbusy.wav inside the path you;ve mentioned. So I believe it is correct. Date: Tue, 3 Feb 2009 19:50:22 -0600 Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? From: ryanstudyvo...@gmail.com To: lovingprin...@gmail.com CC: kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com; gree...@googlemail.com; christian.hennr...@intact-is.com; anil...@yahoo.com Where are the Blocked pattern reason codes greeting stored at? Are they under program filesciscotftpathunited states On Tue, Feb 3, 2009 at 12:46 AM, kamal yousaf lovingprin...@gmail.com wrote: Yeah..I did that but putting DN didn't work.You would need Secondary AC pilot #. Besides, I prefer to use Unity rather than going through this method.At least for lab, it won't be advisable unless strictly asked to do so. On Tue, Feb 3, 2009 at 5:02 PM, Kapil Atrish kapilatr...@hotmail.com wrote: Cool...I did not check for the TCD Service Parameter. I think if I set this parameter the second AC would not be required. I may simply put a DN as Always route member to extend fast busy to caller after initial MOH. Otherwise I'll also follow your solution. Vik/Mark: Do you think it is an acceptable solution? Question is to customize annunciator and we are using MOH to acheive the results? Date: Tue, 3 Feb 2009 15:39:43 +1100 Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? From: lovingprin...@gmail.com To: kapilatr...@hotmail.com CC: anthony.ye...@gmail.com; ccie_voice@onlinestudylist.com; gree...@googlemail.com; christian.hennr...@intact-is.com; anil...@yahoo.com Kapil, If you dial your first AC pilot # , you should hear greeting .If you dial second(dummy) AC pilot # , you should hear user busy.Now, When you link 2nd to first,i.e add 2nd AC pilot # as Always Route Member , after hold time expires, call will be routed to dummy pilot point and you will get 'user busy'.I did also change Service Parameter for TCD so that AC can route calls to directory numbers with unknown state. Regds On Mon, Feb 2, 2009 at 9:25 PM, Kapil Atrish kapilatr...@hotmail.com wrote: Hi Kamal, I created additional AC Pilot with queuing disabled and pointed first one to the new AC as Alwasy Route Member. I keep on getting the MOH from first AC even though queuing timer is over. Can you pl comment if you achieved it differently? I am able to route the call to a CTI_RP as Alwasy Route Member and point this RP to a route-pattern which further points it to the Gateway. The RP string is invalid and non-routable by the GW. Using this method, the caller simply gets dropped after queuing timer is over. No Fast-busy to caller but MOH gets played. When I try pointing AC Always Route Member filed to any Route-pattern directly, I get the following message: The Directory Number you entered in the selected Partition is associated with a device that can not be a member of a Hunt Group. Pl let me know how you achieved Anthony's method? Date: Thu, 29 Jan 2009 18:05:31 +1100 Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? From: lovingprin...@gmail.com To: anthony.ye...@gmail.com CC: kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com; gree...@googlemail.com; christian.hennr...@intact-is.com; anil...@yahoo.com I tested it and it works great.Thanks Anthony for kind help. On Thu, Jan 29, 2009 at 3:52 PM, Anthony Yeung anthony.ye...@gmail.com wrote: What you can try is assign a dummy AC pilot point to the original AC Pilot Point as the 'Always Route Member' creating a linked hunt group. Then for this second dummy AC pilot point assign a dummy AC user like you did w/ the first. But instead, disable queuing for this second dummy Pilot Point. After the hold time expires for the first AC pilot, the call will be forwarded to the second AC pilot. Since queuing is disabled, the call should drop BUT w/ a disconnect cause of 'user busy'. On Wed, Jan 28, 2009 at 11:15 AM, Kapil Atrish kapilatr...@hotmail.com wrote: I did not put the TP directly inside the Hunt-Group. I put a CTIRP as Always Route Member and on CTIRP I did a forward all to the TP. I am yet to try the solution given by Christian. I'll put the call to a gateway via RP and see if I can get fast-busy to the caller after initial queuing prompt. Date: Tue, 27 Jan 2009 21:30:58 +1100 Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? From: lovingprin...@gmail.com To: kapilatr...@hotmail.com CC: christian.hennr...@intact-is.com; gree...@googlemail.com; ccie_voice@onlinestudylist.com; anil...@yahoo.com I tried same way.It plays greeting only once.I also changed service parameter for Cisco TCD Allow Routing with Unknown Line State to True ,and retried.Call still doesn't end. Kapil, how
Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
Hi Kamal, I created additional AC Pilot with queuing disabled and pointed first one to the new AC as Alwasy Route Member. I keep on getting the MOH from first AC even though queuing timer is over. Can you pl comment if you achieved it differently? I am able to route the call to a CTI_RP as Alwasy Route Member and point this RP to a route-pattern which further points it to the Gateway. The RP string is invalid and non-routable by the GW. Using this method, the caller simply gets dropped after queuing timer is over. No Fast-busy to caller but MOH gets played. When I try pointing AC Always Route Member filed to any Route-pattern directly, I get the following message: The Directory Number you entered in the selected Partition is associated with a device that can not be a member of a Hunt Group. Pl let me know how you achieved Anthony's method? Date: Thu, 29 Jan 2009 18:05:31 +1100Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?From: lovingprin...@gmail.comto: anthony.ye...@gmail.comcc: kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com; gree...@googlemail.com; christian.hennr...@intact-is.com; anil...@yahoo.comi tested it and it works great.Thanks Anthony for kind help. On Thu, Jan 29, 2009 at 3:52 PM, Anthony Yeung anthony.ye...@gmail.com wrote: What you can try is assign a dummy AC pilot point to the original ACPilot Point as the 'Always Route Member' creating a linked hunt group.Then for this second dummy AC pilot point assign a dummy AC user likeyou did w/ the first. But instead, disable queuing for this seconddummy Pilot Point. After the hold time expires for the first AC pilot,the call will be forwarded to the second AC pilot. Since queuing isdisabled, the call should drop BUT w/ a disconnect cause of 'userbusy'. On Wed, Jan 28, 2009 at 11:15 AM, Kapil Atrish kapilatr...@hotmail.com wrote: I did not put the TP directly inside the Hunt-Group. I put a CTIRP as Always Route Member and on CTIRP I did a forward all to the TP. I am yet to try the solution given by Christian. I'll put the call to a gateway via RP and see if I can get fast-busy to the caller after initial queuing prompt. Date: Tue, 27 Jan 2009 21:30:58 +1100 Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? From: lovingprin...@gmail.com To: kapilatr...@hotmail.com CC: christian.hennr...@intact-is.com; gree...@googlemail.com; ccie_voice@onlinestudylist.com; anil...@yahoo.com I tried same way.It plays greeting only once.I also changed service parameter for Cisco TCD Allow Routing with Unknown Line State to True ,and retried.Call still doesn't end. Kapil, how did you add TP as member in HuntGroup.In my case, it gives error saying that member should be a valid DN on system.I was able to add phone/CTIRP DNs though. On Tue, Jan 27, 2009 at 8:48 PM, Kapil Atrish kapilatr...@hotmail.com wrote: I tried with RP/TP Block this pattern and in that case call stays in queue. AC takes the call out of the queue only when it is routed to a registered end-point that's what I've observed. I'll try to route it to some unallocated number pointing it to the GW and see if it works. Thanks for the input. Date: Tue, 27 Jan 2009 10:39:31 +0100 From: christian.hennr...@intact-is.com To: kapilatr...@hotmail.com CC: gree...@googlemail.com; anil...@yahoo.com; cpar...@cparker.us; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? what about routing to a number CUCM, which does not exist, or even to a PSTN number, which is unallocated? Christian Kapil Atrish schrieb: The requirement is to drop the call within CCM itself. I don't want to use Unity/IPCCX/TCL for this purpose. Date: Tue, 27 Jan 2009 09:16:49 + Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? From: gree...@googlemail.com To: anil...@yahoo.com CC: christian.hennr...@intact-is.com; cpar...@cparker.us; kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com Folks, To get the call to disconnect you can use do the following: Create a CTI RP cfwd all to voicemail. In VM create a CH with the extension number of the CTI RP and configure the greeting to be blank and then after greeting send the caller to hang up. In the ac hunt group config add the CTI RP as the always route member. In acconfig.bat for the annunicator ac pilot set the hold time to be something other than 0 seconds After this time has passed the call will be forwarded to unity and disconnected - you get a little bit of ringing as the call gets to unity which I cant get rid of. 2009/1/27 anil batra anil...@yahoo.com I too tried the way Kapil mentioned and faced same issue as he did. The call from PSTN does it the announcement but the call never gets disonncted, it seems the queue
Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
Yes you are right Juan but we are not going to login the AC user to console application. It is used just to populate the line group members. I could've added dummy phone DNs instead of AC user and achieved the same result. Always Route Member field is what we are using here and not the AC user/lines. Let me know if it doesn't clarify... Date: Thu, 29 Jan 2009 11:32:27 +0100 Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? From: juan.c...@gmail.com To: lovingprin...@gmail.com CC: anthony.ye...@gmail.com; kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com; christian.hennr...@intact-is.com; anil...@yahoo.com; gree...@googlemail.com I do not understand the part Then for this second dummy AC pilot point assign a dummy AC user like you did w/ the first . These AC users (not the 'ac' user created in DC directory and linked to both AC pilot points) aren't they only used when using the Attendand Client/user-line AC members regards,Juan On Thu, Jan 29, 2009 at 8:05 AM, kamal yousaf lovingprin...@gmail.com wrote: I tested it and it works great.Thanks Anthony for kind help. On Thu, Jan 29, 2009 at 3:52 PM, Anthony Yeung anthony.ye...@gmail.com wrote: What you can try is assign a dummy AC pilot point to the original AC Pilot Point as the 'Always Route Member' creating a linked hunt group. Then for this second dummy AC pilot point assign a dummy AC user like you did w/ the first. But instead, disable queuing for this second dummy Pilot Point. After the hold time expires for the first AC pilot, the call will be forwarded to the second AC pilot. Since queuing is disabled, the call should drop BUT w/ a disconnect cause of 'user busy'. On Wed, Jan 28, 2009 at 11:15 AM, Kapil Atrish kapilatr...@hotmail.com wrote: I did not put the TP directly inside the Hunt-Group. I put a CTIRP as Always Route Member and on CTIRP I did a forward all to the TP. I am yet to try the solution given by Christian. I'll put the call to a gateway via RP and see if I can get fast-busy to the caller after initial queuing prompt. Date: Tue, 27 Jan 2009 21:30:58 +1100 Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? From: lovingprin...@gmail.com To: kapilatr...@hotmail.com CC: christian.hennr...@intact-is.com; gree...@googlemail.com; ccie_voice@onlinestudylist.com; anil...@yahoo.com I tried same way.It plays greeting only once.I also changed service parameter for Cisco TCD Allow Routing with Unknown Line State to True ,and retried.Call still doesn't end. Kapil, how did you add TP as member in HuntGroup.In my case, it gives error saying that member should be a valid DN on system.I was able to add phone/CTIRP DNs though. On Tue, Jan 27, 2009 at 8:48 PM, Kapil Atrish kapilatr...@hotmail.com wrote: I tried with RP/TP Block this pattern and in that case call stays in queue. AC takes the call out of the queue only when it is routed to a registered end-point that's what I've observed. I'll try to route it to some unallocated number pointing it to the GW and see if it works. Thanks for the input. Date: Tue, 27 Jan 2009 10:39:31 +0100 From: christian.hennr...@intact-is.com To: kapilatr...@hotmail.com CC: gree...@googlemail.com; anil...@yahoo.com; cpar...@cparker.us; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? what about routing to a number CUCM, which does not exist, or even to a PSTN number, which is unallocated? Christian Kapil Atrish schrieb: The requirement is to drop the call within CCM itself. I don't want to use Unity/IPCCX/TCL for this purpose. Date: Tue, 27 Jan 2009 09:16:49 + Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? From: gree...@googlemail.com To: anil...@yahoo.com CC: christian.hennr...@intact-is.com; cpar...@cparker.us; kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com Folks, To get the call to disconnect you can use do the following: Create a CTI RP cfwd all to voicemail. In VM create a CH with the extension number of the CTI RP and configure the greeting to be blank and then after greeting send the caller to hang up. In the ac hunt group config add the CTI RP as the always route member. In acconfig.bat for the annunicator ac pilot set the hold time to be something other than 0 seconds After this time has passed the call will be forwarded to unity and disconnected - you get a little bit of ringing as the call gets to unity which I cant get rid of. 2009/1/27 anil batra anil...@yahoo.com I too tried the way Kapil mentioned and faced same issue as he did. The call from PSTN does it the announcement but the call never gets
Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
I did not put the TP directly inside the Hunt-Group. I put a CTIRP as Always Route Member and on CTIRP I did a forward all to the TP. I am yet to try the solution given by Christian. I'll put the call to a gateway via RP and see if I can get fast-busy to the caller after initial queuing prompt. Date: Tue, 27 Jan 2009 21:30:58 +1100 Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? From: lovingprin...@gmail.com To: kapilatr...@hotmail.com CC: christian.hennr...@intact-is.com; gree...@googlemail.com; ccie_voice@onlinestudylist.com; anil...@yahoo.com I tried same way.It plays greeting only once.I also changed service parameter for Cisco TCD Allow Routing with Unknown Line State to True ,and retried.Call still doesn't end. Kapil, how did you add TP as member in HuntGroup.In my case, it gives error saying that member should be a valid DN on system.I was able to add phone/CTIRP DNs though. On Tue, Jan 27, 2009 at 8:48 PM, Kapil Atrish kapilatr...@hotmail.com wrote: I tried with RP/TP Block this pattern and in that case call stays in queue. AC takes the call out of the queue only when it is routed to a registered end-point that's what I've observed. I'll try to route it to some unallocated number pointing it to the GW and see if it works. Thanks for the input. Date: Tue, 27 Jan 2009 10:39:31 +0100 From: christian.hennr...@intact-is.com To: kapilatr...@hotmail.com CC: gree...@googlemail.com; anil...@yahoo.com; cpar...@cparker.us; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? what about routing to a number CUCM, which does not exist, or even to a PSTN number, which is unallocated? Christian Kapil Atrish schrieb: The requirement is to drop the call within CCM itself. I don't want to use Unity/IPCCX/TCL for this purpose. Date: Tue, 27 Jan 2009 09:16:49 + Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? From: gree...@googlemail.com To: anil...@yahoo.com CC: christian.hennr...@intact-is.com; cpar...@cparker.us; kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com Folks, To get the call to disconnect you can use do the following: Create a CTI RP cfwd all to voicemail. In VM create a CH with the extension number of the CTI RP and configure the greeting to be blank and then after greeting send the caller to hang up. In the ac hunt group config add the CTI RP as the always route member. In acconfig.bat for the annunicator ac pilot set the hold time to be something other than 0 seconds After this time has passed the call will be forwarded to unity and disconnected - you get a little bit of ringing as the call gets to unity which I cant get rid of. 2009/1/27 anil batra anil...@yahoo.com I too tried the way Kapil mentioned and faced same issue as he did. The call from PSTN does it the announcement but the call never gets disonncted, it seems the queue is holdin git for forever. Anyone here has tested this and have some workaround please. --- On *Tue, 1/27/09, Kapil Atrish /kapilatr...@hotmail.com/* wrote: From: Kapil Atrish kapilatr...@hotmail.com Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? To: christian.hennr...@intact-is.com, cpar...@cparker.us Cc: ccie_voice@onlinestudylist.com Date: Tuesday, January 27, 2009, 11:38 AM Chris, Your suspicion is what I've in mind that's why I am trying to avoid using Unity/IPCCX/TCL. I've tested AC workaround and its working for me but couple of catches. First of all, the file is in form of MOH and not annunciator which was the original requirement of the question. Secondly, I am not able to disconnect the call. The message keeps on playing until caller drops the call. thanks, Kapil Atrish Date: Mon, 26 Jan 2009 18:57:28 +0100 From: christian.hennr...@intact-is.com To: cpar...@cparker.us CC: ryanstudyvo...@gmail.com; kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? Hi, what about having a MoH File, that is playing the message to the caller. MoH file is played in the AC Hunt group with queueing activated and no AC operators logged in. So you would use only CUCM to play the message. I have not tested that idea, but it might be workable. As far as there is nothing stated, which
Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
Cool... I'll try this as well. I noticed AC takes the call out of queue only only when it has a registered device which can answer the call. Else call remains in queue. But thanks for opening the additional door. I'll def check this one. Date: Wed, 28 Jan 2009 22:52:40 -0600 Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? From: anthony.ye...@gmail.com To: kapilatr...@hotmail.com CC: lovingprin...@gmail.com; ccie_voice@onlinestudylist.com; gree...@googlemail.com; christian.hennr...@intact-is.com; anil...@yahoo.com What you can try is assign a dummy AC pilot point to the original AC Pilot Point as the 'Always Route Member' creating a linked hunt group. Then for this second dummy AC pilot point assign a dummy AC user like you did w/ the first. But instead, disable queuing for this second dummy Pilot Point. After the hold time expires for the first AC pilot, the call will be forwarded to the second AC pilot. Since queuing is disabled, the call should drop BUT w/ a disconnect cause of 'user busy'. On Wed, Jan 28, 2009 at 11:15 AM, Kapil Atrish kapilatr...@hotmail.com wrote: I did not put the TP directly inside the Hunt-Group. I put a CTIRP as Always Route Member and on CTIRP I did a forward all to the TP. I am yet to try the solution given by Christian. I'll put the call to a gateway via RP and see if I can get fast-busy to the caller after initial queuing prompt. Date: Tue, 27 Jan 2009 21:30:58 +1100 Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? From: lovingprin...@gmail.com To: kapilatr...@hotmail.com CC: christian.hennr...@intact-is.com; gree...@googlemail.com; ccie_voice@onlinestudylist.com; anil...@yahoo.com I tried same way.It plays greeting only once.I also changed service parameter for Cisco TCD Allow Routing with Unknown Line State to True ,and retried.Call still doesn't end. Kapil, how did you add TP as member in HuntGroup.In my case, it gives error saying that member should be a valid DN on system.I was able to add phone/CTIRP DNs though. On Tue, Jan 27, 2009 at 8:48 PM, Kapil Atrish kapilatr...@hotmail.com wrote: I tried with RP/TP Block this pattern and in that case call stays in queue. AC takes the call out of the queue only when it is routed to a registered end-point that's what I've observed. I'll try to route it to some unallocated number pointing it to the GW and see if it works. Thanks for the input.Date: Tue, 27 Jan 2009 10:39:31 +0100 From: christian.hennr...@intact-is.com To: kapilatr...@hotmail.com CC: gree...@googlemail.com; anil...@yahoo.com; cpar...@cparker.us; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? what about routing to a number CUCM, which does not exist, or even to a PSTN number, which is unallocated? Christian Kapil Atrish schrieb: The requirement is to drop the call within CCM itself. I don't want to use Unity/IPCCX/TCL for this purpose. Date: Tue, 27 Jan 2009 09:16:49 + Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? From: gree...@googlemail.com To: anil...@yahoo.com CC: christian.hennr...@intact-is.com; cpar...@cparker.us; kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com Folks, To get the call to disconnect you can use do the following: Create a CTI RP cfwd all to voicemail. In VM create a CH with the extension number of the CTI RP and configure the greeting to be blank and then after greeting send the caller to hang up. In the ac hunt group config add the CTI RP as the always route member. In acconfig.bat for the annunicator ac pilot set the hold time to be something other than 0 seconds After this time has passed the call will be forwarded to unity and disconnected - you get a little bit of ringing as the call gets to unity which I cant get rid of. 2009/1/27 anil batra anil...@yahoo.com I too tried the way Kapil mentioned and faced same issue as he did. The call from PSTN does it the announcement but the call never gets disonncted, it seems the queue is holdin git for forever. Anyone here has tested this and have some workaround please. --- On *Tue, 1/27/09, Kapil Atrish /kapilatr...@hotmail.com/* wrote: From: Kapil Atrish kapilatr...@hotmail.com Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? To: christian.hennr...@intact-is.com, cpar...@cparker.us Cc: ccie_voice@onlinestudylist.com Date: Tuesday, January 27, 2009, 11:38 AM Chris, Your suspicion is what I've in mind that's why I am trying to avoid using Unity/IPCCX/TCL. I've tested AC workaround and its working
Re: [OSL | CCIE_Voice] Channels block after upgrade of DSPs.
Hi, Do you've dspfarm and dsp services dspfarm under voice-cards?? Daniel Sobrinho dani...@hotmail.com wrote: Hello, Could please help me with a doubt? I've been made an upgrade of DSPs in my router 2851 to increase the capacity for transcoder and conference bridge. After applied it a number maximum of sessions for both dspfarm profiles, my gateway stopped receiving external calls from pstn and could not get more to do external calls by pstn link. My service provider said that my voice channels were blocked. Does anybody knows like that or tell me if I did something wrong? IOS: c2800nm-adventerprisek9-mz.124-3.bin Before applied the new DSPs: sccp local GigabitEthernet0/0 sccp ccm 10.55.14.6 identifier 2 version 4.1 sccp ccm 10.55.14.1 identifier 1 version 4.1 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate ccm 2 priority 2 associate profile 2 register CFB_FTZ associate profile 1 register XCODE_FTZ ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec gsmfr codec g729r8 codec g729br8 maximum sessions 14 associate application SCCP ! dspfarm profile 2 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 5 associate application SCCP After applied the new DSPs: sccp local GigabitEthernet0/0 sccp ccm 10.55.14.6 identifier 2 version 4.1 sccp ccm 10.55.14.1 identifier 1 version 4.1 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate ccm 2 priority 2 associate profile 2 register CFB_FTZ associate profile 1 register XCODE_FTZ ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec gsmfr codec g729r8 codec g729br8 maximum sessions 18 associate application SCCP ! dspfarm profile 2 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 16 associate application SCCP == #show dspfarm profile 1 Dspfarm Profile Configuration Profile ID = 1, Service = TRANSCODING, Resource ID = 1 Profile Description : Profile Admin State : UP Profile Operation State : ACTIVE Application : SCCP Status : ASSOCIATED Resource Provider : FLEX_DSPRM Status : UP Number of Resource Configured : 18 Number of Resource Available : 18 Codec Configuration Codec : g711ulaw, Maximum Packetization Period : 30 Codec : g711alaw, Maximum Packetization Period : 30 Codec : g729ar8, Maximum Packetization Period : 60 Codec : g729abr8, Maximum Packetization Period : 60 Codec : gsmfr, Maximum Packetization Period : 20 Codec : g729r8, Maximum Packetization Period : 60 Codec : g729br8, Maximum Packetization Period : 60 #show dspfarm profile 1 2 Dspfarm Profile Configuration Profile ID = 2, Service = CONFERENCING, Resource ID = 2 Profile Description : Profile Admin State : UP Profile Operation State : ACTIVE Application : SCCP Status : ASSOCIATED Resource Provider : FLEX_DSPRM Status : UP Number of Resource Configured : 16 Number of Resource Available : 16 Codec Configuration Codec : g711ulaw, Maximum Packetization Period : 30 , Transcoder: Not Required Codec : g711alaw, Maximum Packetization Period : 30 , Transcoder: Not Required Codec : g729ar8, Maximum Packetization Period : 60 , Transcoder: Not Required Codec : g729abr8, Maximum Packetization Period : 60 , Transcoder: Not Required Codec : g729r8, Maximum Packetization Period : 60 , Transcoder: Not Required Codec : g729br8, Maximum Packetization Period : 60 , Transcoder: Not Required #show diag Slot 0: C2851 Motherboard with 2GE and integrated VPN Port adapter, 2 ports Port adapter is analyzed Port adapter insertion time unknown Onboard VPN: FW ver01100200 EEPROM contents at hardware discovery: PCB Serial Number: FOC10292RYB Hardware Revision: 1.0 Top Assy. Part Number: 800-26922-02 Board Revision : A0 Deviation Number : 0 Fab Version : 03 RMA Test History : 00 RMA Number : 0-0-0-0 RMA History : 00 Processor type : 87 Hardware date code : 20060720 Chassis Serial Number: FTX1031A3XD Chassis MAC Address : 0018.b9ce.5558 MAC Address block size : 32 CLEI Code: COM3E00BRA Product (FRU) Number : CISCO2851 Part Number : 73-8480-04 Version Identifier : V03 EEPROM format version 4 EEPROM contents (hex): 0x00: 04 FF C1 8B 46 4F 43 31 30 32 39 32 52 59 42 40 0x10: 03 E9 41 01 00 C0 46 03 20 00 69 2A 02 42 41 30 0x20: 88 00 00 00 00 02 03 03 00 81 00 00 00 00 04 00 0x30: 09 87 83 01 32 1A 30 C2 8B 46 54 58 31 30 33 31 0x40: 41 33 58 44 C3 06 00 18 B9 CE 55 58 43 00 20 C6 0x50: 8A 43 4F 4D 33 45 30 30 42 52 41 CB 8F 43 49 53 0x60: 43 4F 32 38 35 31 20 20 20 20 20 20 82 49
Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
The requirement is to drop the call within CCM itself. I don't want to use Unity/IPCCX/TCL for this purpose. Date: Tue, 27 Jan 2009 09:16:49 + Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? From: gree...@googlemail.com To: anil...@yahoo.com CC: christian.hennr...@intact-is.com; cpar...@cparker.us; kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com Folks, To get the call to disconnect you can use do the following: Create a CTI RP cfwd all to voicemail. In VM create a CH with the extension number of the CTI RP and configure the greeting to be blank and then after greeting send the caller to hang up. In the ac hunt group config add the CTI RP as the always route member. In acconfig.bat for the annunicator ac pilot set the hold time to be something other than 0 seconds After this time has passed the call will be forwarded to unity and disconnected - you get a little bit of ringing as the call gets to unity which I cant get rid of. 2009/1/27 anil batra anil...@yahoo.com I too tried the way Kapil mentioned and faced same issue as he did. The call from PSTN does it the announcement but the call never gets disonncted, it seems the queue is holdin git for forever. Anyone here has tested this and have some workaround please. --- On Tue, 1/27/09, Kapil Atrish kapilatr...@hotmail.com wrote: From: Kapil Atrish kapilatr...@hotmail.com Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? To: christian.hennr...@intact-is.com, cpar...@cparker.us Cc: ccie_voice@onlinestudylist.com Date: Tuesday, January 27, 2009, 11:38 AM Chris, Your suspicion is what I've in mind that's why I am trying to avoid using Unity/IPCCX/TCL. I've tested AC workaround and its working for me but couple of catches. First of all, the file is in form of MOH and not annunciator which was the original requirement of the question. Secondly, I am not able to disconnect the call. The message keeps on playing until caller drops the call. thanks, Kapil Atrish Date: Mon, 26 Jan 2009 18:57:28 +0100 From: christian.hennr...@intact-is.com To: cpar...@cparker.us CC: ryanstudyvo...@gmail.com; kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? Hi, what about having a MoH File, that is playing the message to the caller. MoH file is played in the AC Hunt group with queueing activated and no AC operators logged in. So you would use only CUCM to play the message. I have not tested that idea, but it might be workable. As far as there is nothing stated, which prevents you from using Unity, I would use Unity. Regards Chris Parker schrieb: The only thing that makes me suspicious about using Unity to play the announcement is that this requirement was listed under the Media section. This leads me to believe they want you to use the annunciator. Otherwise wouldn't it be under the Voicemail/Unity section? Regardless I don't think you can do it any other way unless you hairpin the call through Unity to send the call to the annunciator since the VM ports are skinny registrations. Chris Ryan Trauernicht wrote: That is what I thought but I opened a TAC case and they claim you can, but cant figure out how. Thanks, Ryan Trauernicht On Mon, Jan 26, 2009 at 3:21 AM, Juan juan.c...@gmail.com mailto:juan.c...@gmail.com wrote: I remember reading in the SRND that you can only engage the annunciator for SCCP devices if I remember correctly - so not to the PSTN. cheers, Juan On Sun, Jan 25, 2009 at 5:48 PM, Ryan Trauernicht ryanstudyvo...@gmail.com mailto:ryanstudyvo...@gmail.com wrote: Not sure why you are going through all that trouble and not just sending it to unity as a call handler and hang up after message played. I don't know how to play an ANN from a PSTN call, I have engaged TAC and they are still working on it and they can't even figure it out right now. Any ideas? Thanks, Ryan Trauernicht On Sun, Jan 25, 2009 at 3:42 AM, Kapil Atrish kapilatr...@hotmail.com mailto:kapilatr...@hotmail.com wrote: Hi list, Following I did: Create a new MOH Audio Source using AAExtnOutOfService.wav. Prompt available inside Wfavvid folder Create a TP covering all unassigned DNs for example: 11xx, do Called party Xform to 1155 Create a AC Pilot 1155, give any DP say: ANN_PSTN AC Hunt-GroupGive any AC user. No need to login to Attendant Console. Run acconfig.batEnable Queuing Inside DP: ANN_PSTN give User Hold MOH Source as AAExtnOutOfService.wav. Now, whenever you dial any unassigned number withing range 11xx, you'll hear AAExtnOutOfService.wav but the problem is that I am not able to make the PSTN call drop. I tried routing calls to TP inside
Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
I tried with RP/TP Block this pattern and in that case call stays in queue. AC takes the call out of the queue only when it is routed to a registered end-point that's what I've observed. I'll try to route it to some unallocated number pointing it to the GW and see if it works. Thanks for the input. Date: Tue, 27 Jan 2009 10:39:31 +0100 From: christian.hennr...@intact-is.com To: kapilatr...@hotmail.com CC: gree...@googlemail.com; anil...@yahoo.com; cpar...@cparker.us; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? what about routing to a number CUCM, which does not exist, or even to a PSTN number, which is unallocated? Christian Kapil Atrish schrieb: The requirement is to drop the call within CCM itself. I don't want to use Unity/IPCCX/TCL for this purpose. Date: Tue, 27 Jan 2009 09:16:49 + Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? From: gree...@googlemail.com To: anil...@yahoo.com CC: christian.hennr...@intact-is.com; cpar...@cparker.us; kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com Folks, To get the call to disconnect you can use do the following: Create a CTI RP cfwd all to voicemail. In VM create a CH with the extension number of the CTI RP and configure the greeting to be blank and then after greeting send the caller to hang up. In the ac hunt group config add the CTI RP as the always route member. In acconfig.bat for the annunicator ac pilot set the hold time to be something other than 0 seconds After this time has passed the call will be forwarded to unity and disconnected - you get a little bit of ringing as the call gets to unity which I cant get rid of. 2009/1/27 anil batra anil...@yahoo.com I too tried the way Kapil mentioned and faced same issue as he did. The call from PSTN does it the announcement but the call never gets disonncted, it seems the queue is holdin git for forever. Anyone here has tested this and have some workaround please. --- On *Tue, 1/27/09, Kapil Atrish /kapilatr...@hotmail.com/* wrote: From: Kapil Atrish kapilatr...@hotmail.com Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? To: christian.hennr...@intact-is.com, cpar...@cparker.us Cc: ccie_voice@onlinestudylist.com Date: Tuesday, January 27, 2009, 11:38 AM Chris, Your suspicion is what I've in mind that's why I am trying to avoid using Unity/IPCCX/TCL. I've tested AC workaround and its working for me but couple of catches. First of all, the file is in form of MOH and not annunciator which was the original requirement of the question. Secondly, I am not able to disconnect the call. The message keeps on playing until caller drops the call. thanks, Kapil Atrish Date: Mon, 26 Jan 2009 18:57:28 +0100 From: christian.hennr...@intact-is.com To: cpar...@cparker.us CC: ryanstudyvo...@gmail.com; kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? Hi, what about having a MoH File, that is playing the message to the caller. MoH file is played in the AC Hunt group with queueing activated and no AC operators logged in. So you would use only CUCM to play the message. I have not tested that idea, but it might be workable. As far as there is nothing stated, which prevents you from using Unity, I would use Unity. Regards Chris Parker schrieb: The only thing that makes me suspicious about using Unity to play the announcement is that this requirement was listed under the Media section. This leads me to believe they want you to use the annunciator. Otherwise wouldn't it be under the Voicemail/Unity section? Regardless I don't think you can do it any other way unless you hairpin the call through Unity to send the call to the annunciator since the VM ports are skinny registrations. Chris Ryan Trauernicht wrote: That is what I thought but I opened a TAC case and they claim you can, but cant figure out how. Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
Chris, Your suspicion is what I've in mind that's why I am trying to avoid using Unity/IPCCX/TCL. I've tested AC workaround and its working for me but couple of catches. First of all, the file is in form of MOH and not annunciator which was the original requirement of the question. Secondly, I am not able to disconnect the call. The message keeps on playing until caller drops the call. thanks, Kapil Atrish Date: Mon, 26 Jan 2009 18:57:28 +0100 From: christian.hennr...@intact-is.com To: cpar...@cparker.us CC: ryanstudyvo...@gmail.com; kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? Hi, what about having a MoH File, that is playing the message to the caller. MoH file is played in the AC Hunt group with queueing activated and no AC operators logged in. So you would use only CUCM to play the message. I have not tested that idea, but it might be workable. As far as there is nothing stated, which prevents you from using Unity, I would use Unity. Regards Chris Parker schrieb: The only thing that makes me suspicious about using Unity to play the announcement is that this requirement was listed under the Media section. This leads me to believe they want you to use the annunciator. Otherwise wouldn't it be under the Voicemail/Unity section? Regardless I don't think you can do it any other way unless you hairpin the call through Unity to send the call to the annunciator since the VM ports are skinny registrations. Chris Ryan Trauernicht wrote: That is what I thought but I opened a TAC case and they claim you can, but cant figure out how. Thanks, Ryan Trauernicht On Mon, Jan 26, 2009 at 3:21 AM, Juan juan.c...@gmail.com mailto:juan.c...@gmail.com wrote: I remember reading in the SRND that you can only engage the annunciator for SCCP devices if I remember correctly - so not to the PSTN. cheers, Juan On Sun, Jan 25, 2009 at 5:48 PM, Ryan Trauernicht ryanstudyvo...@gmail.com mailto:ryanstudyvo...@gmail.com wrote: Not sure why you are going through all that trouble and not just sending it to unity as a call handler and hang up after message played. I don't know how to play an ANN from a PSTN call, I have engaged TAC and they are still working on it and they can't even figure it out right now. Any ideas? Thanks, Ryan Trauernicht On Sun, Jan 25, 2009 at 3:42 AM, Kapil Atrish kapilatr...@hotmail.com mailto:kapilatr...@hotmail.com wrote: Hi list, Following I did: Create a new MOH Audio Source using AAExtnOutOfService.wav. Prompt available inside Wfavvid folder Create a TP covering all unassigned DNs for example: 11xx, do Called party Xform to 1155 Create a AC Pilot 1155, give any DP say: ANN_PSTN AC Hunt-GroupGive any AC user. No need to login to Attendant Console. Run acconfig.batEnable Queuing Inside DP: ANN_PSTN give User Hold MOH Source as AAExtnOutOfService.wav. Now, whenever you dial any unassigned number withing range 11xx, you'll hear AAExtnOutOfService.wav but the problem is that I am not able to make the PSTN call drop. I tried routing calls to TP inside AC Hunt-GroupAlways Route member is TPTP has Block Pattern --Not working. AAExtnOutOfService.wavkeeps on playing. I tried routing calls to Route-Point (Always Route Member) inside AC Hunt-GroupCTI_RP has Forward all to TPTP has Block Pattern --Not working. AAExtnOutOfService.wavkeeps on playing. I tried routing calls to a registered Phone DN as Always Route MemberForward all to TPTP has Block Pattern --Not working. AAExtnOutOfService.wavkeeps on playing. Can someone help me achieve call drop here without using IPCCX/Unity/TCL? Thanks, Kapil Atrish Rediscover the magic of Windows WIN a Windows Vista laptop Windows mobile phone at www.windowsandme.com http://www.windowsandme.com Try it now! http://www.windowsandme.com __ This email has been scanned by the MessageLabs Email Security
Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
Chris, Your suspicion is what I've in mind that's why I am trying to avoid using Unity/IPCCX/TCL. I've tested AC workaround and its working for me but couple of catches. First of all, the file is in form of MOH and not annunciator which was the original requirement of the question. Secondly, I am not able to disconnect the call. The message keeps on playing until caller drops the call. thanks, Kapil Atrish Date: Mon, 26 Jan 2009 18:57:28 +0100 From: christian.hennr...@intact-is.com To: cpar...@cparker.us CC: ryanstudyvo...@gmail.com; kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? Hi, what about having a MoH File, that is playing the message to the caller. MoH file is played in the AC Hunt group with queueing activated and no AC operators logged in. So you would use only CUCM to play the message. I have not tested that idea, but it might be workable. As far as there is nothing stated, which prevents you from using Unity, I would use Unity. Regards Chris Parker schrieb: The only thing that makes me suspicious about using Unity to play the announcement is that this requirement was listed under the Media section. This leads me to believe they want you to use the annunciator. Otherwise wouldn't it be under the Voicemail/Unity section? Regardless I don't think you can do it any other way unless you hairpin the call through Unity to send the call to the annunciator since the VM ports are skinny registrations. Chris Ryan Trauernicht wrote: That is what I thought but I opened a TAC case and they claim you can, but cant figure out how. Thanks, Ryan Trauernicht On Mon, Jan 26, 2009 at 3:21 AM, Juan juan.c...@gmail.com mailto:juan.c...@gmail.com wrote: I remember reading in the SRND that you can only engage the annunciator for SCCP devices if I remember correctly - so not to the PSTN. cheers, Juan On Sun, Jan 25, 2009 at 5:48 PM, Ryan Trauernicht ryanstudyvo...@gmail.com mailto:ryanstudyvo...@gmail.com wrote: Not sure why you are going through all that trouble and not just sending it to unity as a call handler and hang up after message played. I don't know how to play an ANN from a PSTN call, I have engaged TAC and they are still working on it and they can't even figure it out right now. Any ideas? Thanks, Ryan Trauernicht On Sun, Jan 25, 2009 at 3:42 AM, Kapil Atrish kapilatr...@hotmail.com mailto:kapilatr...@hotmail.com wrote: Hi list, Following I did: Create a new MOH Audio Source using AAExtnOutOfService.wav. Prompt available inside Wfavvid folder Create a TP covering all unassigned DNs for example: 11xx, do Called party Xform to 1155 Create a AC Pilot 1155, give any DP say: ANN_PSTN AC Hunt-GroupGive any AC user. No need to login to Attendant Console. Run acconfig.batEnable Queuing Inside DP: ANN_PSTN give User Hold MOH Source as AAExtnOutOfService.wav. Now, whenever you dial any unassigned number withing range 11xx, you'll hear AAExtnOutOfService.wav but the problem is that I am not able to make the PSTN call drop. I tried routing calls to TP inside AC Hunt-GroupAlways Route member is TPTP has Block Pattern --Not working. AAExtnOutOfService.wavkeeps on playing. I tried routing calls to Route-Point (Always Route Member) inside AC Hunt-GroupCTI_RP has Forward all to TPTP has Block Pattern --Not working. AAExtnOutOfService.wavkeeps on playing. I tried routing calls to a registered Phone DN as Always Route MemberForward all to TPTP has Block Pattern --Not working. AAExtnOutOfService.wavkeeps on playing. Can someone help me achieve call drop here without using IPCCX/Unity/TCL? Thanks, Kapil Atrish Rediscover the magic of Windows WIN a Windows Vista laptop Windows mobile phone at www.windowsandme.com http://www.windowsandme.com Try it now! http://www.windowsandme.com __ This email has been scanned by the MessageLabs Email Security
[OSL | CCIE_Voice] Annunciator to PSTN - will it be acceptable?
Hi list, Following I did: Create a new MOH Audio Source using AAExtnOutOfService.wav. Prompt available inside Wfavvid folder Create a TP covering all unassigned DNs for example: 11xx, do Called party Xform to 1155 Create a AC Pilot 1155, give any DP say: ANN_PSTN AC Hunt-GroupGive any AC user. No need to login to Attendant Console. Run acconfig.batEnable Queuing Inside DP: ANN_PSTN give User Hold MOH Source as AAExtnOutOfService.wav. Now, whenever you dial any unassigned number withing range 11xx, you'll hear AAExtnOutOfService.wav but the problem is that I am not able to make the PSTN call drop. I tried routing calls to TP inside AC Hunt-GroupAlways Route member is TPTP has Block Pattern --Not working. AAExtnOutOfService.wavkeeps on playing. I tried routing calls to Route-Point (Always Route Member) inside AC Hunt-GroupCTI_RP has Forward all to TPTP has Block Pattern --Not working. AAExtnOutOfService.wavkeeps on playing. I tried routing calls to a registered Phone DN as Always Route MemberForward all to TPTP has Block Pattern --Not working. AAExtnOutOfService.wavkeeps on playing. Can someone help me achieve call drop here without using IPCCX/Unity/TCL? Thanks, Kapil Atrish _ Chose your Life Partner! Join MSN Matrimony FREE http://www.in.msn.com/matrimony
[OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
Hi list, Following I did: Create a new MOH Audio Source using AAExtnOutOfService.wav. Prompt available inside Wfavvid folder Create a TP covering all unassigned DNs for example: 11xx, do Called party Xform to 1155 Create a AC Pilot 1155, give any DP say: ANN_PSTN AC Hunt-GroupGive any AC user. No need to login to Attendant Console. Run acconfig.batEnable Queuing Inside DP: ANN_PSTN give User Hold MOH Source as AAExtnOutOfService.wav. Now, whenever you dial any unassigned number withing range 11xx, you'll hear AAExtnOutOfService.wav but the problem is that I am not able to make the PSTN call drop. I tried routing calls to TP inside AC Hunt-GroupAlways Route member is TPTP has Block Pattern --Not working. AAExtnOutOfService.wavkeeps on playing. I tried routing calls to Route-Point (Always Route Member) inside AC Hunt-GroupCTI_RP has Forward all to TPTP has Block Pattern --Not working. AAExtnOutOfService.wavkeeps on playing. I tried routing calls to a registered Phone DN as Always Route MemberForward all to TPTP has Block Pattern --Not working. AAExtnOutOfService.wavkeeps on playing. Can someone help me achieve call drop here without using IPCCX/Unity/TCL? Thanks, Kapil Atrish _ Plug in to the MSN Tech channel for a full update on the latest gizmos that made an impact. http://computing.in.msn.com/
Re: [OSL | CCIE_Voice] CME BACD - drop-through not working
Hi Narinder, Your configuration worked for me. After having another look at the same cisco doc I realised in all the examples of drop-through there is only single param aa-hunt-grps 1 under AA TCL script. I had aa-hunt-grps 2 which seems invalid in drop-through scenario. thanks for all the help you provided. thanks, Kapil Atrish From: narinder.ku...@uxcg.com.auto: kapilatr...@hotmail.com; ccie_vo...@onlinestudylist.comdate: Mon, 19 Jan 2009 23:39:09 +1100Subject: RE: [OSL | CCIE_Voice] CME BACD - drop-through not working Kapil, This is the configuration which is working on my router no issues, try it. application service callq flash:app-b-acd-2.1.0.0.tcl param queue-len 10 param aa-hunt1 3020 param number-of-hunt-grps 1 param queue-manager-debugs 1 ! service aa flash:app-b-acd-aa-2.1.0.0.tcl paramspace english index 0 param number-of-hunt-grps 1 param drop-through-option 1 param handoff-string aa paramspace english language en param max-time-vm-retry 2 param aa-pilot 3500 paramspace english location flash: param second-greeting-time 60 param drop-through-prompt _dt_prompt.au param call-retry-timer 15 param voice-mail 3600 param max-time-call-retry 700 param service-name callq From: Kapil Atrish [mailto:kapilatr...@hotmail.com] Sent: Monday, 19 January 2009 11:35 PMTo: Kumar, Narinder; ccie_vo...@onlinestudylist.comsubject: RE: [OSL | CCIE_Voice] CME BACD - drop-through not working Thanks Narinder for quick reply, I tried the flash: option but same results, not working.paramspace english location tftp://172.30.1.4/ --- I tried with flash: option with no luck.I am checking attached cisco doc for configuring drop-through. Thanks,Kapil Atrish From: narinder.ku...@uxcg.com.auto: kapilatr...@hotmail.com; ccie_vo...@onlinestudylist.comdate: Mon, 19 Jan 2009 22:50:08 +1100Subject: RE: [OSL | CCIE_Voice] CME BACD - drop-through not working Kapil, I am hoping the bacd script is in router flash. I have always done drop through with a single HG, not sure you can achieve multiple HG”s with drop through ( I could be wrong need to double check. Change service callq tftp://172.30.1.10/app-b-acd-2.1.0.0.tcl toservice callq flash:app-b-acd-2.1.0.0.tcl Change service dropthruaa tftp://172.30.1.10/app-b-acd-aa-2.1.0.0.tclto service dropthruaa flash:app-b-acd-aa-2.1.0.0.tcl Also ur tftp path in paramspace english location tftp://172.30.1.4/ is different to other is it just a typo or some other reason behind this. Thanks Narinder From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kapil AtrishSent: Monday, 19 January 2009 9:34 PMTo: ccie_vo...@onlinestudylist.comsubject: [OSL | CCIE_Voice] CME BACD - drop-through not working Hi, I've CME 3.3 (supports drop-through). I've configured BACD with drop-through functionality but its not working the desired way. When I dial the pilot no: I get silence. If I press 2 or 3 the call get routed to the respective hunt-group. Can someone pl suggest what can I check/change? I've attached:CME Config,dir flash:,Output of SHOW CALL APPLICATION SESSION when script is active,output of debug voice application script, I've reloaded the router but no good. MY TCL scripts are being read successfully from TFTP Server (another question, Is it mandatory in drop-through that TCL script must be accessed via TFTP? I've checked cisco docs which always shows TCL scripts accessed from TFTP whenver drop-through scenario is discussed). Thanks in advance... Make sure your wardrobe reflects the latest trends and styles in the world of fashion. Try it! CONFIDENTIALITY - The information contained in this electronic mail message is confidential and is intended solely for the addressee(s). If you are not an authorised recipient of this message please contact Getronics Australia immediately by reply email and destroy/delete this message from your computer. Any unauthorised form of reproduction of this message, or part thereof, is strictly prohibited.DISCLAIMER - Unless specifically indicated otherwise, the views and opinions expressed in this email are those of the sender and not Getronics Australia. While we endeavour to protect our network from computer viruses, Getronics Australia does not warrant that this email or any attachments are free of viruses or any other defects or errors. It is the duty of the recipient to virus scan and otherwise test any information contained in this email before loading onto any computer system. News, views and chilling images. If it matters for India, we bring it to your fingertips. Check it out! CONFIDENTIALITY - The information contained in this electronic mail message is confidential and is intended solely for the addressee(s). If you are not an authorised recipient of this message please contact Getronics Australia immediately by reply email and destroy/delete
Re: [OSL | CCIE_Voice] B-ACD
Yes ReadMe is available in router flash. I've used it. But be aware it doesn't cover each and every parameter (for ex. drop-through). DocCD is no more provided in the Lab. It has been replaced with following URL: http://www.cisco.com/web/psa/products/index.html kamal yousaf lovingprin...@gmail.com wrote: No need to memorize anything.If readme file is available,then its good.Otherwise,you DocCD and copy-paste script for your use. Here is the navigation Structure: DocCD: Voice and Unified Communications -- IP Telephony Call Control --CUCME -- Configuration Guidelines -- Cisco Unified CME B-ACD and Tcl Call-Handling Applications Rgds On Thu, Jan 22, 2009 at 3:47 PM, Greg Hauser ghau...@cox.net wrote: Hi I have been spending a lot of time with B-ACD and was wondering if anyone knew if CME flash with aap-b-acd-2.1.0.0.tcl readme will be made available to us or do we need to memorize the B-ACD config? Thanks Greg Hauser
Re: [OSL | CCIE_Voice] IPCC script change
Nope. I've done it manier times, you don't need to restart anything. Mike O mik...@msn.com wrote: When you change to a different script in IPCC do you need to restart any services? Thanks, Mike
[OSL | CCIE_Voice] AC-Broadcast hunting not working
Hi, Can someone pl suggest how to debug this issue? I am still not able to make it work? Thanks... From: kapilatr...@hotmail.comto: ccie_vo...@onlinestudylist.comsubject: AC-Broadcast hunting not workingDate: Mon, 22 Dec 2008 22:43:43 +0530 Hi list,I've configured Attendant Console and is working fine for Longest Available/Circular Hunting but not in Broadcast Hunting mode. I've line members and users in Line Group. When system is configured for LAA/Circular Hunting, AC user can see call coming in inside the AC console, but when broadcast hunting is configured the calling party gets MOH from the DP of AC Pilot (User hold Audio Source). AC User cannot see call inside the AC console. I've tried line group with only users (no phones), phones + users but same result. Restart of CCM didn't resolve.Any help is highly appreciated. _ Find a better job. We have plenty. Visit MSN Jobs http://www.in.msn.com/jobs
Re: [OSL | CCIE_Voice] Announciator messages to PSTN
It won't be a normal call even if annunciator answers the PSTN call (firstly it doesn't) because difference in Call Clearing Cause Code issued by GW/CCM for unallocated number, number busy or Call cleared normally etc.. Chris Parker [EMAIL PROTECTED] wrote: Sounds correct to me. I guess the interesting part is that if you did have something in call manager to match the number (like a CTI route point with a line number ) and then forward it to a translation pattern that has call block checked, why wont the annunciator play that message - Your call cannot be completed as dialed? All I have heard is that it just doesn't work. I realize of course from the ISDN perspective, it would just look like a completed call answered by the annunciator. Christian Hennrich wrote: Hi, as far as I know all messages with the appropriate ISDN disconnect causes are played by the network provider itself and not from any PBX. So if you would like to play message to the PSTN, you need Unity, because CCM will send the ISDN disconnect cause and not play any message. But if you have a translation or catch all route pattern in CCM for not available numbers. Then you are able to send the call to unity, where you play a message to the caller. But you need to be aware, that the ISDN code is like a normal call. I would therefore also think that any file manipulation will not help, because CCM sends the ISDN disconnect cause. Please correct me, if I wrong Regards Chris Parker schrieb: So does that mean that the annunciator will play messages to a call coming from Unity but not from a gateway? Or does it mean that Unity is used to play a recorded message to the caller in place of the annunciator? I've seen this question about annunciator pop up a few times and the answer always seems to be to send it to Unity. I'm just trying to understand what exactly Unity does in this scenario. I have also heard the annunciator will only play to SCCP and MGCP devices but not H323. If that is the case then if you have an MGCP gateway like the 6608 and IOS MGCP shouldn't the annunciator work with them? Thanks Chris Hardesty, Scott wrote: You can not use annunciator for pstn. You need to route the call to unity and use call handler... Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio 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: Michael Shavrov Sent: Tuesday, November 18, 2008 11:56 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Announciator messages to PSTN Hi, How to play messages with announciator to PSTN? For example, if PSTN phone calls number, which belongs to location but has no configured DN, user should hear message Number is not in service. I tried to configure both, Route Pattern and Translation Pattern with Block pattern - it works internally, but does not work from PSTN. Also, there is no configurable option for Number not in service - call manager just rejects the call. Mike __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __
Re: [OSL | CCIE_Voice] AAR simple question. How to use AAR in scenarios other that TEHO?
Put Site A phones in AAR Group say SiteA, location SiteA. Put Site B phones in AAR Group say SiteB, location SiteB. Set AAR prefix, AAR CSS and route-patterns. Lower down the b/w and AAR triggers. jeremy co [EMAIL PROTECTED] wrote: Hi, assuming following scenario: _ WAN ___ | | 3XXX -SITE A-- SITE B--- 4XXX | | - PSTN - Centralized model SITE A : CCM if TEHO is to be implemented ,AAR locations can be set on GWs of SITE A and SITE B. e.g calls to site B through WAN exceed defined BW then use siteA's GW to route calls to pstn by AAR . if CME on siteB or GK implemented ,it's easy to run this, and location can be put on GWs. But what happened if we want to put BW restriction for site B 4XXX phones? e.g site B has 10 phones and we have only 96Kbps , so we want just 4 calls go through wan and the fifth call should route by pstn from site A to site B. in above scenario there is no gateway to put location on it since 4xxx are site numbers, not pstn numbers so they route WITHIN callmanager. Any suggestion where to apply location concept to use AAR in above scenario? Jeremy
Re: [OSL | CCIE_Voice] Question regarding leaving Overhead.
Is your question about overhead calculation for voice calls or something else? My explanation for calculating overhead for voice calls. Example: CIR 512, Allow priority b/w for 5 g.729 calls with FRF.12 (can be MLPPP, FR: Allow 10% overhead. I would calculate 5x 27.2kbps = 136kbps Add 10% ovehead = 13.6 Total = 149.6 Kbps. Round-off 150 Kbps You may already be knowing all this. I want to emphasize on the point that overhead is calculated on b/w required for number of calls. I haven't seen any scenario which says leave 10% interface/CIR b/w for overheads. Scott ODonnell [EMAIL PROTECTED] wrote: I've been working through several WAN Qos scenarios and I keep getting hung up in how to interpret requirements. On one hand you have the max-reserved-bandwidth command, which is applied at the physical interface. Then you have the well-established rule of calculating 95% of CIR's for frame sub-interfaces. Given a vague requirement of leaving 10% for overhead, how would you approach this? Do you raise the max-reserved-bandwidth of the physical interface? Or do you adjust the calculated CIR/MINCIR, etc. I know Ask the proctor is the obvious answer. Just looking for input. - Scott
Re: [OSL | CCIE_Voice] Transcoders required for IPCC Exp
Possible your transcoder is not getting invoked. You've HQ region set to use G.711 within itself and G.729 with others. I believe CTI RP and ports would be in HQ region and IPCC Express is configured for G.711 codec. So you get fast-busy. When you change the region settings to use G.711 you can make calls successfully. You may want to recheck xcoder registration, mapping to MRG MRGL and applied to DP or device level. Pardeep Singh (pardsing) [EMAIL PROTECTED] wrote: Hello, I have a question regarding 6608 xcoder for IPCC. I have my 6608 Transcoder in a HQ region which does g711 within and g729 to others. One of my phone in SiteB1 calls the CTI RP for IPCC and gets a fast busy.. If I change my 6608 xcoder to be in all g711 region then the call works fine but its a g711 call from siteb1 phone to ipcc which means I am running g711 over WAN = no good. Can someone shine some light on the proper configuration needed for this. Thank you in advance.
Re: [OSL | CCIE_Voice] Block calling name
On route-pattern you may set CLID Name/Number to restricted/allowed. James Key [EMAIL PROTECTED] wrote: Block calling nameWhat is the best way to block calling name on certain route patterns, while still allowing it on others? Example: hq local send calling name + number, hq international just calling number. thanks, James NOTICE: This electronic mail message and any files transmitted with it are intended exclusively for the individual or entity to which it is addressed. The message, together with any attachment, may contain confidential and/or privileged information. Any unauthorized review, use, printing, saving, copying, disclosure or distribution is strictly prohibited. If you have received this message in error, please immediately advise the sender by reply email and delete all copies.
Re: [OSL | CCIE_Voice] cme phones to two different unity systems
translation-profile incoming on ephone-dn and translate the voicemail number to CUE or Unity Pilot. Leave other phone without translation. Balamurugan Singaram [EMAIL PROTECTED] wrote: Hi, We have two cme phones in BR2 two different unity systems: 1st phone press messages button and go to unity 4.0.5 greetings 2nd phone press messages button and go to CUE greetings How to make it work? Thanks, - New Email names for you! Get the Email name you've always wanted on the new @ymail and @rocketmail. Hurry before someone else does!
Re: [OSL | CCIE_Voice] Gatekeeper E164 registration
I ran into this problem number of times. I initially put number without no-reg option under ephone-dn and when integrating it with GK later I simply put the command number no-reg: and I faced this issue. I need to do no number and number no-reg to resolve this. Same way for ephone-hunt Pilot number, dialplan-pattern number. Kumar, Narinder [EMAIL PROTECTED] wrote: RE: [OSL | CCIE_Voice] Gatekeeper E164 registrationWhen you do no telephony setup, and telephony service again by default the ephones will register back as no reg/no reg both wont be in the config . Do reset telephony service all or best bet is to reload the CME router. No gateway sometime doesnt fix the problem From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Key Sent: Thursday, 13 November 2008 1:27 PM To: Greg Miglucci (gmiglucc); ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Gatekeeper E164 registration Thanks Greg. Did that (several times actually) also shut down the gatekeeper, did a no gateway on cme and did a no telephony setup and started over. Still same issue. I was very frustrated. Cost me valuable time and points as I was never able to resolve. -Original Message- From: Greg Miglucci (gmiglucc) [mailto:[EMAIL PROTECTED] Sent: Wed 11/12/2008 7:10 PM To: James Key; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Gatekeeper E164 registration Verify no-reg and then do no gateway gateway on the CME router. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Key Sent: Wednesday, November 12, 2008 4:45 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Gatekeeper E164 registration Had an issue where all my DNs on cme would register to Gatekeeper. I had no-reg defined for each number, and it still would always register those numbers. Doing a no gateway and then gateway never resolved. Did I miss something somewhere? I never have run into this issue during my studies. The one thing I didn't do was reboot the gatekeeper router. James NOTICE: This electronic mail message and any files transmitted with it are intended exclusively for the individual or entity to which it is addressed. The message, together with any attachment, may contain confidential and/or privileged information. Any unauthorized review, use, printing, saving, copying, disclosure or distribution is strictly prohibited. If you have received this message in error, please immediately advise the sender by reply email and delete all copies. NOTICE: This electronic mail message and any files transmitted with it are intended exclusively for the individual or entity to which it is addressed. The message, together with any attachment, may contain confidential and/or privileged information. Any unauthorized review, use, printing, saving, copying, disclosure or distribution is strictly prohibited. If you have received this message in error, please immediately advise the sender by reply email and delete all copies. - CONFIDENTIALITY - The information contained in this electronic mail message is confidential and is intended solely for the addressee(s). If you are not an authorised recipient of this message please contact Getronics Australia immediately by reply email and destroy/delete this message from your computer. Any unauthorised form of reproduction of this message, or part thereof, is strictly prohibited. DISCLAIMER - Unless specifically indicated otherwise, the views and opinions expressed in this email are those of the sender and not Getronics Australia. While we endeavour to protect our network from computer viruses, Getronics Australia does not warrant that this email or any attachments are free of viruses or any other defects or errors. It is the duty of the recipient to virus scan and otherwise test any information contained in this email before loading onto any computer system.
Re: [OSL | CCIE_Voice] IPCC lab Gotcha
You may need to upload the welcome-prompt under respective language (en_US, en, Default). Did you validate the script before uploading under CRS Script editor? You may've run the script in reactive mode and verify whether the call was hitting the script or not and which stage it was failing. Steve ccietester [EMAIL PROTECTED] wrote: Just made another lab attempt recently and result was not good. One of the things that really made me frustrated is IPCC express. I couldn't get it working in either of my two attempts. When i dialed CTI RP number(from either HQ or branch site), I got no sound and then busy signal after a few seconds. In both attempts I checked that CTI RP was registered, partitions and CSS were properly assigned, transcoder was registered and placed in the right MRGL. Everything seemed to be correctly configured but call just couldn't get through. I felt pretty confident on ipcc express before going into lab as I have been practising it in my own lab for at least 30 times and I knew to what to check. Failure twice on the same thing is really scary. So I think maybe there is a gotcha in the real lab that I'm not aware of (e.g service parameter)? Has anyone experienced same problem when taking the lab? What else do I need to look? Thanks a lot!!
Re: [OSL | CCIE_Voice] Finally !, CCIE Voice #22488 ...from Chile
100% correct. I had the same issue and I had to remove the dialplan pattern and use TP under voice-port to meet the requirement of 4 digit CLID to HQ 10 digit DID to PSTN . I've made it a practice not to use dialplan pattern. However one small confusion when having BACD. For ex Question says: aa-pilot no should be 3223000. Now, if you use TP under voice-port and translate all incoming calls to 4 digit extension (DID), the your aa-pilot would be 3000 and not 3223000. I don't know how would proctor grade that since question asks aa-pilot to be 3000. There is a wayaround to use num-exp and expand 3000 to 3223000 but I never tested this. May be Christian can share his experience on this as well... Christian Narvaez [EMAIL PROTECTED] wrote: RE: [OSL | CCIE_Voice] Finally !, CCIE Voice #22488 ...from ChileHi Chunmei, In the real exam it is asked a lot of requirements related with transformations of ANI or DNIS for the CME and SRST. My experience in prior attempts say that is not recommendable using dial-plan pattern, instead of get used to use translation rules/profile and apply then to the dial-peer to accomplish each specific question requirement. For example imagine it is asked that phones in CME need to be seen as a 4 digits ANI when calling to HQ or SiteB, in that case you will have problem using dial-plan pattern since is force you to send the 10 digits ANI, same for access to CUE, or in case is required all international calls from CME present ANI with a preempted 9011. In those cases anyway you will need to use some kind of translations In resumen I thinks is more flexible using Translations than dial-plan pattern in relation to the kind of questions asked in the exam. -Original Message- From: chunmei chen [mailto:[EMAIL PROTECTED] Sent: Sat 11/1/2008 7:18 AM To: Christian Narvaez Subject: Re: [OSL | CCIE_Voice] Finally !, CCIE Voice #22488 ...from Chile Congra Christian!! way to go! Your notes below is very helpful.. however could you explain a little bit more on avoiding dialplan pattern? In what scenario it causes conflict? I have been using dialplan pattern command since day one never had a problem. Thanks! --- On Thu, 10/30/08, Christian Narvaez [EMAIL PROTECTED] wrote: From: Christian Narvaez [EMAIL PROTECTED] Subject: [OSL | CCIE_Voice] Finally !, CCIE Voice #22488 ...from Chile To: ccie_voice@onlinestudylist.com Date: Thursday, October 30, 2008, 8:46 AM I am glad to announce this October 27th I finally obtained my CCIE Voice #22488. I would like to thank all the people of this forum, especially those who shed my path when I was lost in some topics. Special thanks to Mark and Vik and the team of IPexpert which support this forum Below is the detail of my experience and the thoughts I would like to share with other candidates. Some Facts of my preparation Number of attempts: 4 Attempt Dates : May 5th, June 26th , August 20th and October 27th Location Center: All in San Jose Months of preparations: 12 since passed the written test. Hours working on Virtual Rack Sessions: 416 Hours (52 IPExpert Proctorlabs session, 8 hours/each) Hours working on own lab : aprox 800 Hours checking written material and forums : aprox 400 Books Read: 0 ,is not needed if you are not a beginner Bootcamps attended : 0 , although it depends of each one, but personally I think is costly in relation with the real benefit. Forums Consulted: Internetwork Expert (web-based) and IPExpert (email distribution), both are good. Cost per attempt: aprox USD 3200 (Exam=USD 1400 , AirTicket from CHILE=USD 1500, Stay+Transportation+Food=USD 300~600) Strategies Used during my attempts -- Strategy 1) Section-Based Approach, Configuring and Testing the whole section before begin the next. I had a predefined amount of max time for each section that I could afford to complete the configuration and testing before go on with the next section. Strategy 2) At the beginning of the test spend max 20 minutes doing the strategy3 and a brief read to just some key questions specially the one of the LocationCAC Section Strategy 3) Cut the large paper sheet given in San Jose into four smaller pieces a) One of the pieces for the topologic diagram, IP Addresses and Numbering Plan b) On the second piece, write down each section name and the task numbers. b.1)Besides each section name, note the max estimated time when you expect finish the section, that is useful to self-control the time you spend specially if a problem is faced. b.2) Once configured the task mark it with a check besides b.3) Once tested the task mark it with an OK besides. c) On the third piece write down the numbers of the PSTN IP Phone, believe me this simple tip saves time when you are testing dial plan and you will not have to moving
[OSL | CCIE_Voice] Annunciator on unassigned number
HI, Not sure if it has been asked and answered earlier. Requirement, play CCM annunciator on incoming call from PSTN to any unassigned DID. For ex DID range 200-300. Ext 250 to 300 are not assigned to any device. If PSTN calls any of these DIDs, the caller gets fast-busy. Instead of fast-busy I want to play CCM annunciator saying something like number not available. It works for internal callers without any additional config. I tried a TP with Block this parttern, unassigned number etc options but no luck. Tried creating a CTI RP covering the unassigned number. CRI TP remains unregistered. Same result, fast-busy to caller. I can achieve this by routing calls to Unity and playing required prompt but that's not the requirement. Questions says CCM Annunciator needs to be played and not Unity greeting. Thanks for your help.. _ Searching for weekend getaways? Try Live.com http://www.live.com/?scope=videoform=MICOAL
Re: [OSL | CCIE_Voice] Gateway Channel selection control ???
When adding CAS circuit in CCM (MGCP) it gives option to enable channels for Outbound/Inbound/Bothways. I never tried but won't that work for us. Secondly, if CAS circuit in H.323 mode you may create multiple DS0 and point DPs to respective voice-ports for outbound calls leaving others for inbound. Paul and Bobs [EMAIL PROTECTED] wrote: Thanks for the reply. How can this this be achieved. I know from the PABX world , you can control the number of outbound and inbound channels and would love to find a way of doing this i the Cisco world on either MGCP or H.323. Doesnt matter which one (both would be good) but if not then just one of them. Cheers On Wed, Oct 29, 2008 at 11:02 PM, Mark Snow [EMAIL PROTECTED] wrote: No because you can't create two pri-group timeslot service mgcp commands because then you would have to create two mgcp gateways in CUCM and 4.1.3 won't allow you to do this to the same hostname. Mark SnowSr Technical Instructor IPexpert, Inc. Sent from my iPhone On Oct 28, 2008, at 11:36 PM, Paul and Bobs [EMAIL PROTECTED] wrote: Thanks Mark If I wanted to just use mgcp, is there a way to control which channels are used. So I can reserve 10 channeles for outgoing and 10 for incoming with a total of 20 On Wed, Oct 29, 2008 at 2:19 PM, Mark Snow [EMAIL PROTECTED] wrote: BTW - that's not to say that I recommend it - but for lab purposes should be all good. There could be bugs associated with it - and I would definitely check BugNavigator before putting it into production :) cheers, -- Mark Snow CCIE #14073 (Voice, Security) Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: [EMAIL PROTECTED] -- Join our free online support and peer group communities: http://www.IPexpert.com/communities -- IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- On Oct 28, 2008, at 10:20 PM, Paul and Bobs wrote: Hi All I was wandering if anyone know of a way using both MGCP and H.323 to control the channells on an E1/T1 circuit. For example - If I have a single E1 service with only 20 channels and I want to say reserve 5 for outgoing and reserve 15 for incoming, is there a way on both protocols to do this. Thanks Paul
Re: [OSL | CCIE_Voice] Where to run IPMA
Isn't the call processing depends upon which CCM (Sub/Pub) the IP Phone/IPMA
CTI Port is registering to? If Phone/CTI RP are registered to Sub, all calls
will be processed by Sub even though IPMA points to Pub.
Correct me if I am missing something..
Yung Hung [EMAIL PROTECTED] wrote:v\:* {behavior:url(#default#VML);}
o\:* {behavior:url(#default#VML);} w\:* {behavior:url(#default#VML);} .shape
{behavior:url(#default#VML);} Juan,
I believe you would use DNS and a DNS host to point to both IPs, that way it
will use whichever http server that is up and running.
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Juan
Sent: Tuesday, October 28, 2008 6:46 AM
To: [EMAIL PROTECTED]
Cc: Mark Snow; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Where to run IPMA
Hi Mark, Michael, Bala,
w.r.t the service URL itself, to have the manager be able to change any
setting to the service when the primary http server is down - is it an idea to
create 2 service URLs - a primary IPMA pointing to the primary and a 'backup'
IPMA pointing to the backup http server? Or is this not done for some reason?
regards,
Juan
On Tue, Oct 28, 2008 at 5:41 AM, Balamurugan Singaram [EMAIL PROTECTED]
wrote:
Hi Mark,
If Subscriber will be the Primary Call Processing Server then the
service URL IP Address of IPMA should be only SUB IP address, not PUB IP
Address could please let me know ?
Thanks,
Bala.
--- On Mon, 27/10/08, Mark Snow [EMAIL PROTECTED] wrote:
From: Mark Snow [EMAIL PROTECTED]
Subject: Re: [OSL | CCIE_Voice] Where to run IPMA
To: Michael Shavrov [EMAIL PROTECTED]
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Date: Monday, 27 October, 2008, 9:57 PM
All functionality will continue to work with the exception that if
the manager went to change any preferences, s/he would have to wait until the
http server came back online or phone up the asst and have them do it.
Mark Snow
Sr Technical Instructor
IPexpert, Inc.
Sent from my iPhone
On Oct 27, 2008, at 12:19 PM, Michael Shavrov [EMAIL PROTECTED] wrote:
Mark,
And what about IP service? It's understood, that it's possible to run
IPMA service and configure both in Service Parameters. But how manager's
phone will react on inability to access URL for the IP service?
Sincerely,
Mike
- Original Message -
From: Mark Snow
To: Kevin Porter
Cc: ccie_voice@onlinestudylist.com
Sent: Monday, October 27, 2008 11:50 AM
Subject: Re: [OSL | CCIE_Voice] Where to run IPMA
Point it to both as primary Sub and Backup Pub. IPMA supports both a
pri and sec.
Mark Snow
Sr Technical Instructor
IPexpert, Inc.
Sent from my iPhone
On Oct 27, 2008, at 11:38 AM, Kevin Porter [EMAIL PROTECTED] wrote:
In a scenario where you are told that the Subscriber will be the
Primary Call Processing Server and the Publisher the Backup, should the
IPMA parameters (Phone Service URL, IPMA Service parameters, etc
) point to
the Subscribers IP Address?
Thanks,
Kevin
Kevin Porter
Systems Engineer L4
Netelligent Corporation
400 South Woods Mill Drive, Suite 105
St. Louis , MO 63017
Office: (314) 392-6921
Cell: (314) 852-1252
Fax: (314) 392-9760
[EMAIL PROTECTED]
www.netelligent.com
Bridging The Gap Between Good and GREAT IP Communications!
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[OSL | CCIE_Voice] codec sampling rate
Does the codec sampling rate need to match at CCM and H.323 GW or whatever configured at CCM H.323 GW auto-negotiates? What if different codec sampling rate at CME, does CCM also need to have the same sampling rate? Appreciate any comments on this _ Searching for weekend getaways? Try Live.com http://www.live.com/?scope=videoform=MICOAL
[OSL | CCIE_Voice] CUE - no prompt when dialing Pilot no.
Hi, I've configured CUE/CME and I need few phones to test. My softphones fail to register with CUE. I've tried wit IP Blue and IPC which keeps on registering/unregistering. I've verified loads are present in flash and CME is configured as TFTP for all the load files. I am doing manual registration but no sucess. Any inputs how to fix this. Secondly any debugs to confirm whether my transcoder is getting invoked at CME? When I dial from HQ phone, and the call goes CFNA, I see HQ phone getting redirected to cue VM pilot but don't get any message/prompt. Transcoder is configured on CME and allow-connections h323 to sip and vice-versa. Below are the dial-peers I've configured on CME: VM Pilot 3111, I am using 10 digit dial-plan pattern under telephony-system. 8000/8001 are MWI on and Off nos. Phone 1- 3001, phone-2 3002 having mailboxes configured. ! dial-peer voice 11 voip destination-pattern 3111 session protocol sipv2 session target ipv4:10.3.202.20 dtmf-relay sip-notify codec g711ulaw no vad ! dial-peer voice 12 voip destination-pattern 3...$ session protocol sipv2 session target ipv4:10.3.202.1 incoming called-number 800. dtmf-relay sip-notify codec g711ulaw no vad ! dial-peer voice 13 voip destination-pattern 3313233 session protocol sipv2 session target ipv4:10.3.202.20 dtmf-relay sip-notify codec g711ulaw no vad ! dial-peer voice 14 voip destination-pattern 3313233001 Mailbox 1, ext 3001. session protocol sipv2 session target ipv4:10.3.202.20 dtmf-relay sip-notify codec g711ulaw no vad ! dial-peer voice 15 voip destination-pattern 3313233002mailbox2, ext 3002 session protocol sipv2 session target ipv4:10.3.202.20 dtmf-relay sip-notify codec g711ulaw no vad ! _ Movies, sports news! Get your daily entertainment fix, only on live.com http://www.live.com/?scope=videoform=MICOAL
Re: [OSL | CCIE_Voice] debug multicast MOH
One quick question: Although I've it configured on sub-if and virtual-template, when doing MLPP fragmentation, I need to put ip pim-dense mode only on int virtual-template and not on the sub-if? From: [EMAIL PROTECTED] To: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] debug multicast MOH Date: Sat, 11 Oct 2008 19:05:13 +0530 No I didn't have ccm-manager music-on-hold command. I've put it now. When I do debug ip igmp, I can see message exchange the moment I put the phone on hold. Counters also start increasing in show ip pim interface count at both the routers. Show ip mroute also shows expected output. I've registered couple of IP Blue sofphones at BR2 deivice pool and one IPC at HQ device pool for testing. I am not sure if this scenario is good enough to confirm Multicast MOH b'coz none of the phone is across the wan. CC: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] debug multicast MOH Date: Fri, 10 Oct 2008 12:28:20 -0700 Yes it is- ccm-manager music is required for MGCP AND (I repeat AND) H323 gateways. Vik Malhi – CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On- Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. On Oct 10, 2008, at 12:25 PM, Jacob Owen wrote: did you run the ccm-manager music-on-hold command on the BR1 router? I have been told that is required for Multicast MOH even when you aren't doing MOH from the routers flash. Hopefully someone can chime in and confirm. MSN Technology brings you the latest on gadgets, gizmos and the new hits in the gaming market. Try it now! _ Want to explore the world? Visit MSN Travel for the best deals. http://in.msn.com/coxandkings
[OSL | CCIE_Voice] debug multicast MOH
Hi, Can pl let me know any debug commands to confirm multicast MOH reaching BR1 Router (or Phones if possible). I checked through perfomance monitor on CCM and could see one MOH Multicast resource active when call was put on hold but there was no MOH on phone. I checked show ip pim interface count and could see counters increasing on HQ router interfaces but not on BR1. Below is the summary how I configured multicast MOH: Enabled MOH Audio resource for multicast and MOH server for multicast with Hop count of 6. Codec G.711ulaw an d G.729 selected. Put MOH server in two different MRGs, one for BR1 and anther for HQ. BR1 MRG has MOH enabled. Put the MRGs in respective MRGLs and applied to correct device pools. Enabled IGMP snooping on Cat 6k, enabled multicast-routing on HQ and BR1 routers. Configured IP pim-dense mode on HQ - to BR1 Sub-interface, HQ Fast-ethernet sub-interface, BR1 Voice vlan interface. Thanks for your time.. _ Search for videos of Bollywood, Hollywood, Mollywood and every other wood, only on Live.com http://www.live.com/?scope=videoform=MICOAL
[OSL | CCIE_Voice] SIP call fails on G729
Hi, Following scenario: IP Phone ---CME---sip trunk---CCM--IP Phone Using g729 call fails and works fine on 711. MTP is selected on trunk, infact I've created an IOS enchanced software MTP on a router and given it to SIP trunk. Bt that's software only and I understand it would support only G711. Tried with CCM SW Mtp but same result. Can pl comment if that's the case why calls are failing on 729 or what might be missing. SIP dial-peer config: ! dial-peer voice 9 voip destination-pattern [1-2]...$ session protocol sipv2 session target ipv4:10.5.0.1 dtmf-relay sip-notify codec g711ulaw no vad Thanks for your time.. _ Search for videos of Bollywood, Hollywood, Mollywood and every other wood, only on Live.com http://www.live.com/?scope=videoform=MICOAL
Re: [OSL | CCIE_Voice] SIP call fails on G729
That;s what I am trying to find out if I need the Xcoder. I think if I've xcoder the call would work even w/o ipip gw. Is Xcoder must in this scenario? Thanks for your time... Date: Mon, 6 Oct 2008 04:51:56 -0700 From: [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] SIP call fails on G729 To: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com try to terminate your call from CME to IPIPGW (w/transocder) on HQ router, don't direct to CCM. so the topology will looks like this IP Phone -- CME -- SIP/H323 Trunk -- IPIPGW -- SIP/H323 trunk -- CCM -- IP Phone - Original Message From: Kapil Atrish [EMAIL PROTECTED] To: ccie_voice@onlinestudylist.com Sent: Monday, October 6, 2008 6:47:23 PM Subject: [OSL | CCIE_Voice] SIP call fails on G729 Hi, Following scenario: IP Phone ---CME---sip trunk---CCM--IP Phone Using g729 call fails and works fine on 711. MTP is selected on trunk, infact I've created an IOS enchanced software MTP on a router and given it to SIP trunk. Bt that's software only and I understand it would support only G711. Tried with CCM SW Mtp but same result. Can pl comment if that's the case why calls are failing on 729 or what might be missing. SIP dial-peer config: ! dial-peer voice 9 voip destination-pattern [1-2]...$ session protocol sipv2 session target ipv4:10.5.0.1 dtmf-relay sip-notify codec g711ulaw no vad Thanks for your time.. MSN Technology brings you the latest on gadgets, gizmos and the new hits in the gaming market. Try it now! _ Movies, sports news! Get your daily entertainment fix, only on live.com http://www.live.com/?scope=videoform=MICOAL
Re: [OSL | CCIE_Voice] BACD issue - No welcome prompt
Exactly that was the case. Not it works like charm. Thanks a ton.. CC: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] BACD issue - No welcome prompt Date: Sat, 4 Oct 2008 12:52:37 -0700 I know. Please do as I state in previous email. Any call using the loopback address in a voip dialpeer will still require gk authorization even though we know it is a local call. Vik Malhi – CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] Join our free online support and peer group communities: http://www.IPexpert.com/communitiesIPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. On Oct 4, 2008, at 11:43 AM, Kapil Atrish wrote:Its local call from CME phone to bacd. No gatekeeper in between. PH1---CME with AA/ACD---ephone=hunt CC: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] BACD issue - No welcome prompt Date: Sat, 4 Oct 2008 11:36:07 -0700 Do a debug RAS and check if you see an ARJ from the gatekeeper. Try unregistering the CME from the GK and try. Any call that uses a VOIP dialpeer will require bandwidth authorization for 128kbps and if you have a bandwidth cac restriction within the cme zone the call will fail. Vik Malhi – CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] Join our free online support and peer group communities: http://www.IPexpert.com/communitiesIPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. On Oct 4, 2008, at 10:00 AM, Kapil Atrish wrote:The attached file has full config and debug output if you wish to see. ! dial-peer voice 15 voip destination-pattern 3700 session target ipv4:172.22.102.1 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 16 voip service aa incoming called-number 3700 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! Thanks for your time Subject: RE: [OSL | CCIE_Voice] BACD issue - No welcome prompt Date: Sat, 4 Oct 2008 12:39:38 -0400 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com Can you send your dial-peer for the BACDapplication? Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | [EMAIL PROTECTED]: 301.313.2041 | C: 443.789.1219 | www.presidio.com From: [EMAIL PROTECTED]:[EMAIL PROTECTED] On Behalf Of Kapil Atrish Sent: Saturday, October 04, 20087:58 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] BACDissue - No welcome prompt HI, Attached is my config. I get fast busy tone and Unknown number on display whenI dial the pilot number from any CME phone. I can dial hunt-pilot directly andcall get routed correctly or give the aa-pilot to hunt-pilot and ring thephones fine. Call in between phones are setup using G711ulaw. I've tried singlevoip dial-peer with incoming called-address and destination-pattern, reload ofrouter, re-configure script. Below is the snapshot of bacd config and debug voice application seesion.. application service queue flash:app-b-acd-2.1.0.0.tcl param queue-len 15 param aa-hunt5 3701 param queue-manager-debugs 1 param number-of-hunt-grps 2 ! service aa flash:app-b-acd-aa-2.1.0.0.tcl paramspace english index 1 param number-of-hunt-grps 2 param menu-timeout 6 param handoff-string aa param dial-by-extension-option 4 paramspace english language en param max-time-vm-retry 2 param max-extension-length 4 param aa-pilot 3700 paramspace english location flash: param second-greeting-time 30 param welcome-prompt _bacd_welcome.au param queue-manager-debugs 1 param call-retry-timer 15 param max-time-call-retry 600 param voice-mail 3005 paramspace english prefix en param service-name queue ! ! ! BR2#dir flash: Directory of flash:/ 1 -rw- 24679 no date app-b-acd-2.1.0.0.tcl 2 -rw- 33870 no date app-b-acd-aa-2.1.0.0.tcl 3 -rw- 75650 no date en_bacd_allagentsbusy.au 4 -rw- 83291 no date en_bacd_disconnect.au 5 -rw- 63055 no date en_bacd_enter_dest.au 6 -rw- 37952 no date en_bacd_invalidoption.au 7 -rw- 496521 no date en_bacd_music_on_hold.au 8 -rw- 123446 no date
Re: [OSL | CCIE_Voice] SIP call fails on G729
I don't have hw resources on the router. All I configured is he enchaned IOS MTP which is an IOS feature. AFAN it can only support G711. Which even CCM SW MTP does. Since MTP is a must on SIP trunk and I am running g729 which is not supported by Software MTPs, I a pretty sure now I need the hw resources. Protocol translation can be done by IPIPGW or even CCM. I'll check Cisco design guides also today to confirm. I've excluded CCM MTP from the MRGL bt same result. Thanks for your time.. Subject: RE: [OSL | CCIE_Voice] SIP call fails on G729Date: Mon, 6 Oct 2008 09:49:50 -0400From: [EMAIL PROTECTED]: [EMAIL PROTECTED]; [EMAIL PROTECTED]; [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com You would need to ensure that the Hardware transcoder is used first. Do not have the software and hardware in the same MRG and the MRG with the software must be listed below the hardware MRG Cheers! Mark Cardwell | Systems Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | [EMAIL PROTECTED] D: 571.225.0132 | www.presidio.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael ShavrovSent: Monday, October 06, 2008 8:54 AMTo: Edi Hamlet; Kapil Atrish; [EMAIL PROTECTED]: Re: [OSL | CCIE_Voice] SIP call fails on G729 So. may be we should exclude the software MTP from the MRGL, and keep the only hardware MTP/XCoder? - Original Message - From: Edi Hamlet To: Kapil Atrish ; [EMAIL PROTECTED] Sent: Monday, October 06, 2008 8:04 AM Subject: Re: [OSL | CCIE_Voice] SIP call fails on G729 in order to join 2 different call leg (sip h323) with g729 codec, i think transcoder (MTP hardware)is a must. MTP software can do this but only for g711 codec. - Original Message From: Kapil Atrish [EMAIL PROTECTED]To: Edi Hamlet [EMAIL PROTECTED]; [EMAIL PROTECTED]: Monday, October 6, 2008 6:57:13 PMSubject: RE: [OSL | CCIE_Voice] SIP call fails on G729That;s what I am trying to find out if I need the Xcoder. I think if I've xcoder the call would work even w/o ipip gw. Is Xcoder must in this scenario?Thanks for your time... Date: Mon, 6 Oct 2008 04:51:56 -0700From: [EMAIL PROTECTED]: Re: [OSL | CCIE_Voice] SIP call fails on G729To: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com try to terminate your call from CME to IPIPGW (w/transocder) on HQ router, don't direct to CCM. so the topology will looks like thisIP Phone -- CME -- SIP/H323 Trunk -- IPIPGW -- SIP/H323 trunk -- CCM -- IP Phone - Original Message From: Kapil Atrish [EMAIL PROTECTED]To: [EMAIL PROTECTED]: Monday, October 6, 2008 6:47:23 PMSubject: [OSL | CCIE_Voice] SIP call fails on G729Hi,Following scenario:IP Phone ---CME---sip trunk---CCM--IP PhoneUsing g729 call fails and works fine on 711. MTP is selected on trunk, infact I've created an IOS enchanced software MTP on a router and given it to SIP trunk. Bt that's software only and I understand it would support only G711. Tried with CCM SW Mtp but same result.Can pl comment if that's the case why calls are failing on 729 or what might be missing.SIP dial-peer config:!dial-peer voice 9 voipdestination-pattern [1-2]...$ session protocol sipv2 session target ipv4:10.5.0.1dtmf-relay sip-notify codec g711ulaw no vadThanks for your time.. MSN Technology brings you the latest on gadgets, gizmos and the new hits in the gaming market. Try it now! Voice your opinion on the burning issues of the day. Discuss, debate with the world. Logon to message boards on MSN. Try it! _ Search for videos of Bollywood, Hollywood, Mollywood and every other wood, only on Live.com http://www.live.com/?scope=videoform=MICOAL
[OSL | CCIE_Voice] Cat6K T1 Ports fail to register
HI, I am getting following when trying to register Cat6K port to CCM on Pod 20. I've tried enabling/disabling the ports but all three ports (T1, Xcode and CFB) are in same state. Reset the DHCP service, other devices are able to take IP Address from DHCP and enough IPs are available in the scope. Can't clear CDP table due to insufficient privileges. ort Name Status Vlan Duplex Speed Type - -- -- -- --- 7/4 POD20-PSTN-T1enabled400 full - unknown Port DHCPMAC-Address IP-Address Subnet-Mask --- - --- --- 7/4 enable 00-d0-c0-d3-12-c3 (Failed to obtain port interface information) Appreciate any help... _ Searching for weekend getaways? Try Live.com http://www.live.com/?scope=videoform=MICOAL
[OSL | CCIE_Voice] FW: BACD issue - No welcome prompt
debug voice ccapi inout - using two dial-peers debug voice ccapi inout - using single dial-peer. Result is same, Unknown number, fast-busy tone. From: [EMAIL PROTECTED] To: ccie_voice@onlinestudylist.com Subject: BACD issue - No welcome prompt Date: Sat, 4 Oct 2008 17:27:31 +0530 HI, Attached is my config. I get fast busy tone and Unknown number on display when I dial the pilot number from any CME phone. I can dial hunt-pilot directly and call get routed correctly or give the aa-pilot to hunt-pilot and ring the phones fine. Call in between phones are setup using G711ulaw. I've tried single voip dial-peer with incoming called-address and destination-pattern, reload of router, re-configure script. Below is the snapshot of bacd config and debug voice application seesion.. application service queue flash:app-b-acd-2.1.0.0.tcl param queue-len 15 param aa-hunt5 3701 param queue-manager-debugs 1 param number-of-hunt-grps 2 ! service aa flash:app-b-acd-aa-2.1.0.0.tcl paramspace english index 1 param number-of-hunt-grps 2 param menu-timeout 6 param handoff-string aa param dial-by-extension-option 4 paramspace english language en param max-time-vm-retry 2 param max-extension-length 4 param aa-pilot 3700 paramspace english location flash: param second-greeting-time 30 param welcome-prompt _bacd_welcome.au param queue-manager-debugs 1 param call-retry-timer 15 param max-time-call-retry 600 param voice-mail 3005 paramspace english prefix en param service-name queue ! ! ! BR2#dir flash: Directory of flash:/ 1 -rw- 24679no date app-b-acd-2.1.0.0.tcl 2 -rw- 33870no date app-b-acd-aa-2.1.0.0.tcl 3 -rw- 75650no date en_bacd_allagentsbusy.au 4 -rw- 83291no date en_bacd_disconnect.au 5 -rw- 63055no date en_bacd_enter_dest.au 6 -rw- 37952no date en_bacd_invalidoption.au 7 -rw- 496521no date en_bacd_music_on_hold.au 8 -rw- 123446no date en_bacd_options_menu.au 9 -rw- 42978no date en_bacd_welcome.au 10 -rw- 496521 Mar 01 2002 01:13:09 +00:01 music-on-hold_3db.au 11 -rw- 496521 Mar 01 2002 02:47:07 +00:01 music-on-hold.au 536870908 bytes total (534895700 bytes free) BR2# BR2# There is no output when I do debug voice application script OUTPUT OF 'debug voice application session' is as below. Calling no: 3002, called no: 3700 BR2#debug voice app BR2#debug voice application sess voip application session debugging is on BR2# Mar 1 03:17:40: //37//AFW_:/Closing_AnyEvent: Mar 1 03:17:40: //37//AFW_:/Session_Cleaner: Mar 1 03:17:40: //-1//AFW_:/C_ServiceSession_Event_Handler: Mar 1 03:17:40: //37/8A066DCF802F/AFW_:/C_ServiceSession_Event_Handler: Received event CC_EV_CALL_DISCONNECT_DONE[17] in Main Loop Mar 1 03:17:40: //37//AFW_:/AFW_M_Session_Terminate: Mar 1 03:17:40: //-1//AFW_:HN000FF5F4:/AFW_M_Session_Free: MOD[Session_65BFE164_0_1046004]( ) Mar 1 03:17:42: //-1//AFW_:/C_ServiceSession_Event_Handler: Mar 1 03:17:42: //-1//AFW_:/AFW_Session_New: Mar 1 03:17:42: //40//AFW_:/C_PackageSession_NewCall: Session module listened by TclModule_65BE268C_0_1066356 Mar 1 03:17:42: //40//AFW_:/Open_SetupIndication: Calling #(3002), Called #(), peer_tag(20002) Mar 1 03:17:44: //40//AFW_:/GettingDest_DigitCollectDone: status(4) discCause(0) ovrlp(TRUE) Mar 1 03:17:44: //-1//AFW_:/C_PackageSession_GetSigPeer: Mar 1 03:17:44: //40//AFW_:/ContactingDest_SetupDone: Mar 1 03:17:44: //40//AFW_:/Session_Close: lastFailureCause 34 Mar 1 03:17:44: //40//AFW_:/AFW_M_Session_Terminate: Mar 1 03:17:44: //40//AFW_:/AFW_M_Session_Terminate: lastFailureCause 34 Mar 1 03:17:44: //40//AFW_:/Session_Cleaner: Mar 1 03:17:47: //40//AFW_:/Closing_AnyEvent: Mar 1 03:17:47: //40//AFW_:/Session_Cleaner: Mar 1 03:17:47: //-1//AFW_:/C_ServiceSession_Event_Handler: Mar 1 03:17:47: //40/96270EF58032/AFW_:/C_ServiceSession_Event_Handler: Received event CC_EV_CALL_DISCONNECT_DONE[17] in Main Loop Mar 1 03:17:47: //40//AFW_:/AFW_M_Session_Terminate: Mar 1 03:17:47: //-1//AFW_:HN00104580:/AFW_M_Session_Free: MOD[Session_65BFE214_0_1066368]( ) BR2# BR2# BR2# BR2# Any inputs are very welcome... MSN Technology brings you the latest on gadgets, gizmos and the new hits in the gaming market. Try it now! _ Searching for weekend getaways? Try Live.com http://www.live.com/?scope=videoform=MICOALBR2#sh run Building configuration... Current configuration : 6065 bytes ! ! No configuration change since last restart ! version 12.4 service timestamps debug datetime localtime service timestamps log datetime localtime no service password-encryption ! hostname BR2 !
[OSL | CCIE_Voice] BACD issue - No welcome prompt
HI, Attached is my config. I get fast busy tone and Unknown number on display when I dial the pilot number from any CME phone. I can dial hunt-pilot directly and call get routed correctly or give the aa-pilot to hunt-pilot and ring the phones fine. Call in between phones are setup using G711ulaw. I've tried single voip dial-peer with incoming called-address and destination-pattern, reload of router, re-configure script. Below is the snapshot of bacd config and debug voice application seesion.. application service queue flash:app-b-acd-2.1.0.0.tcl param queue-len 15 param aa-hunt5 3701 param queue-manager-debugs 1 param number-of-hunt-grps 2 ! service aa flash:app-b-acd-aa-2.1.0.0.tcl paramspace english index 1 param number-of-hunt-grps 2 param menu-timeout 6 param handoff-string aa param dial-by-extension-option 4 paramspace english language en param max-time-vm-retry 2 param max-extension-length 4 param aa-pilot 3700 paramspace english location flash: param second-greeting-time 30 param welcome-prompt _bacd_welcome.au param queue-manager-debugs 1 param call-retry-timer 15 param max-time-call-retry 600 param voice-mail 3005 paramspace english prefix en param service-name queue ! ! ! BR2#dir flash: Directory of flash:/ 1 -rw- 24679no date app-b-acd-2.1.0.0.tcl 2 -rw- 33870no date app-b-acd-aa-2.1.0.0.tcl 3 -rw- 75650no date en_bacd_allagentsbusy.au 4 -rw- 83291no date en_bacd_disconnect.au 5 -rw- 63055no date en_bacd_enter_dest.au 6 -rw- 37952no date en_bacd_invalidoption.au 7 -rw- 496521no date en_bacd_music_on_hold.au 8 -rw- 123446no date en_bacd_options_menu.au 9 -rw- 42978no date en_bacd_welcome.au 10 -rw- 496521 Mar 01 2002 01:13:09 +00:01 music-on-hold_3db.au 11 -rw- 496521 Mar 01 2002 02:47:07 +00:01 music-on-hold.au 536870908 bytes total (534895700 bytes free) BR2# BR2# There is no output when I do debug voice application script OUTPUT OF debug voice application session is as below. Calling no: 3002, called no: 3700 BR2#debug voice app BR2#debug voice application sess voip application session debugging is on BR2# Mar 1 03:17:40: //37//AFW_:/Closing_AnyEvent: Mar 1 03:17:40: //37//AFW_:/Session_Cleaner: Mar 1 03:17:40: //-1//AFW_:/C_ServiceSession_Event_Handler: Mar 1 03:17:40: //37/8A066DCF802F/AFW_:/C_ServiceSession_Event_Handler: Received event CC_EV_CALL_DISCONNECT_DONE[17] in Main Loop Mar 1 03:17:40: //37//AFW_:/AFW_M_Session_Terminate: Mar 1 03:17:40: //-1//AFW_:HN000FF5F4:/AFW_M_Session_Free: MOD[Session_65BFE164_0_1046004]( ) Mar 1 03:17:42: //-1//AFW_:/C_ServiceSession_Event_Handler: Mar 1 03:17:42: //-1//AFW_:/AFW_Session_New: Mar 1 03:17:42: //40//AFW_:/C_PackageSession_NewCall: Session module listened by TclModule_65BE268C_0_1066356 Mar 1 03:17:42: //40//AFW_:/Open_SetupIndication: Calling #(3002), Called #(), peer_tag(20002) Mar 1 03:17:44: //40//AFW_:/GettingDest_DigitCollectDone: status(4) discCause(0) ovrlp(TRUE) Mar 1 03:17:44: //-1//AFW_:/C_PackageSession_GetSigPeer: Mar 1 03:17:44: //40//AFW_:/ContactingDest_SetupDone: Mar 1 03:17:44: //40//AFW_:/Session_Close: lastFailureCause 34 Mar 1 03:17:44: //40//AFW_:/AFW_M_Session_Terminate: Mar 1 03:17:44: //40//AFW_:/AFW_M_Session_Terminate: lastFailureCause 34 Mar 1 03:17:44: //40//AFW_:/Session_Cleaner: Mar 1 03:17:47: //40//AFW_:/Closing_AnyEvent: Mar 1 03:17:47: //40//AFW_:/Session_Cleaner: Mar 1 03:17:47: //-1//AFW_:/C_ServiceSession_Event_Handler: Mar 1 03:17:47: //40/96270EF58032/AFW_:/C_ServiceSession_Event_Handler: Received event CC_EV_CALL_DISCONNECT_DONE[17] in Main Loop Mar 1 03:17:47: //40//AFW_:/AFW_M_Session_Terminate: Mar 1 03:17:47: //-1//AFW_:HN00104580:/AFW_M_Session_Free: MOD[Session_65BFE214_0_1066368]( ) BR2# BR2# BR2# BR2# Any inputs are very welcome... _ Search for videos of Bollywood, Hollywood, Mollywood and every other wood, only on Live.com http://www.live.com/?scope=videoform=MICOALBR2#sh run Building configuration... Current configuration : 6065 bytes ! ! No configuration change since last restart ! version 12.4 service timestamps debug datetime localtime service timestamps log datetime localtime no service password-encryption ! hostname BR2 ! boot-start-marker boot-end-marker ! enable secret 5 $1$Z.S8$wuVpMYCLqLqyhpcpHG6eC1 ! no aaa new-model ! resource policy ! memory-size iomem 5 clock timezone CET 0 1 clock summer-time CST recurring ip subnet-zero ip cef ! ! ! ! no ip domain lookup ! ! ! ! ! ! voice service voip allow-connections h323 to h323 ! ! ! ! ! ! ! ! ! ! voice
Re: [OSL | CCIE_Voice] BACD issue - No welcome prompt
The attached file has full config and debug output if you wish to see. ! dial-peer voice 15 voip destination-pattern 3700 session target ipv4:172.22.102.1 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 16 voip service aa incoming called-number 3700 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! Thanks for your time Subject: RE: [OSL | CCIE_Voice] BACD issue - No welcome prompt Date: Sat, 4 Oct 2008 12:39:38 -0400 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com Can you send your dial-peer for the BACDapplication? Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | [EMAIL PROTECTED]: 301.313.2041 | C: 443.789.1219 | www.presidio.com From: [EMAIL PROTECTED]:[EMAIL PROTECTED] On Behalf Of Kapil Atrish Sent: Saturday, October 04, 20087:58 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] BACDissue - No welcome prompt HI, Attached is my config. I get fast busy tone and Unknown number on display whenI dial the pilot number from any CME phone. I can dial hunt-pilot directly andcall get routed correctly or give the aa-pilot to hunt-pilot and ring thephones fine. Call in between phones are setup using G711ulaw. I've tried singlevoip dial-peer with incoming called-address and destination-pattern, reload ofrouter, re-configure script. Below is the snapshot of bacd config and debug voice application seesion.. application service queue flash:app-b-acd-2.1.0.0.tcl param queue-len 15 param aa-hunt5 3701 param queue-manager-debugs 1 param number-of-hunt-grps 2 ! service aa flash:app-b-acd-aa-2.1.0.0.tcl paramspace english index 1 param number-of-hunt-grps 2 param menu-timeout 6 param handoff-string aa param dial-by-extension-option 4 paramspace english language en param max-time-vm-retry 2 param max-extension-length 4 param aa-pilot 3700 paramspace english location flash: param second-greeting-time 30 param welcome-prompt _bacd_welcome.au param queue-manager-debugs 1 param call-retry-timer 15 param max-time-call-retry 600 param voice-mail 3005 paramspace english prefix en param service-name queue ! ! ! BR2#dir flash: Directory of flash:/ 1 -rw- 24679 no date app-b-acd-2.1.0.0.tcl 2 -rw- 33870 no date app-b-acd-aa-2.1.0.0.tcl 3 -rw- 75650 no date en_bacd_allagentsbusy.au 4 -rw- 83291 no date en_bacd_disconnect.au 5 -rw- 63055 no date en_bacd_enter_dest.au 6 -rw- 37952 no date en_bacd_invalidoption.au 7 -rw- 496521 no date en_bacd_music_on_hold.au 8 -rw- 123446 no date en_bacd_options_menu.au 9 -rw- 42978 no date en_bacd_welcome.au 10 -rw- 496521 Mar 012002 01:13:09 +00:01 music-on-hold_3db.au 11 -rw- 496521 Mar 012002 02:47:07 +00:01 music-on-hold.au 536870908 bytes total (534895700 bytes free) BR2# BR2# There is no output when I do debug voice application script OUTPUT OF 'debug voice application session' is as below. Calling no: 3002, called no: 3700 BR2#debug voice app BR2#debug voice application sess voip application session debugging is on BR2# Mar 1 03:17:40: //37//AFW_:/Closing_AnyEvent: Mar 1 03:17:40: //37//AFW_:/Session_Cleaner: Mar 1 03:17:40: //-1//AFW_:/C_ServiceSession_Event_Handler: Mar 1 03:17:40: //37/8A066DCF802F/AFW_:/C_ServiceSession_Event_Handler:Received event CC_EV_CALL_DISCONNECT_DONE[17] in Main Loop Mar 1 03:17:40: //37//AFW_:/AFW_M_Session_Terminate: Mar 1 03:17:40: //-1//AFW_:HN000FF5F4:/AFW_M_Session_Free: MOD[Session_65BFE164_0_1046004]( ) Mar 1 03:17:42: //-1//AFW_:/C_ServiceSession_Event_Handler: Mar 1 03:17:42: //-1//AFW_:/AFW_Session_New: Mar 1 03:17:42: //40//AFW_:/C_PackageSession_NewCall: Session module listenedby TclModule_65BE268C_0_1066356 Mar 1 03:17:42: //40//AFW_:/Open_SetupIndication: Calling #(3002), Called#(), peer_tag(20002) Mar 1 03:17:44: //40//AFW_:/GettingDest_DigitCollectDone: status(4)discCause(0) ovrlp(TRUE) Mar 1 03:17:44: //-1//AFW_:/C_PackageSession_GetSigPeer: Mar 1 03:17:44: //40//AFW_:/ContactingDest_SetupDone: Mar 1 03:17:44: //40//AFW_:/Session_Close: lastFailureCause 34 Mar 1 03:17:44: //40//AFW_:/AFW_M_Session_Terminate: Mar 1 03:17:44: //40//AFW_:/AFW_M_Session_Terminate: lastFailureCause 34 Mar 1 03:17:44: //40//AFW_:/Session_Cleaner: Mar 1 03:17:47: //40//AFW_:/Closing_AnyEvent: Mar 1 03:17:47: //40//AFW_:/Session_Cleaner: Mar 1 03:17:47: //-1//AFW_:/C_ServiceSession_Event_Handler: Mar 1 03:17:47: //40/96270EF58032/AFW_:/C_ServiceSession_Event_Handler:Received event CC_EV_CALL_DISCONNECT_DONE[17] in Main Loop Mar 1 03:17:47: //40//AFW_:/AFW_M_Session_Terminate: Mar 1 03:17:47
Re: [OSL | CCIE_Voice] BACD issue - No welcome prompt
Its local call from CME phone to bacd. No gatekeeper in between. PH1---CME with AA/ACD---ephone=hunt Vikram Malhi [EMAIL PROTECTED] wrote: Do a debug RAS and check if you see an ARJ from the gatekeeper. Try unregistering the CME from the GK and try. Any call that uses a VOIP dialpeer will require bandwidth authorization for 128kbps and if you have a bandwidth cac restriction within the cme zone the call will fail. Vik Malhi CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. On Oct 4, 2008, at 10:00 AM, Kapil Atrish wrote: The attached file has full config and debug output if you wish to see. ! dial-peer voice 15 voip destination-pattern 3700 session target ipv4:172.22.102.1 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 16 voip service aa incoming called-number 3700 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! Thanks for your time - Subject: RE: [OSL | CCIE_Voice] BACD issue - No welcome prompt Date: Sat, 4 Oct 2008 12:39:38 -0400 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com Can you send your dial-peer for the BACDapplication? Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | [EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | www.presidio.com - From: [EMAIL PROTECTED]:[EMAIL PROTECTED] On Behalf Of Kapil Atrish Sent: Saturday, October 04, 20087:58 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] BACDissue - No welcome prompt HI, Attached is my config. I get fast busy tone and Unknown number on display whenI dial the pilot number from any CME phone. I can dial hunt-pilot directly andcall get routed correctly or give the aa-pilot to hunt-pilot and ring thephones fine. Call in between phones are setup using G711ulaw. I've tried singlevoip dial-peer with incoming called-address and destination-pattern, reload ofrouter, re-configure script. Below is the snapshot of bacd config and debug voice application seesion.. application service queue flash:app-b-acd-2.1.0.0.tcl param queue-len 15 param aa-hunt5 3701 param queue-manager-debugs 1 param number-of-hunt-grps 2 ! service aa flash:app-b-acd-aa-2.1.0.0.tcl paramspace english index 1 param number-of-hunt-grps 2 param menu-timeout 6 param handoff-string aa param dial-by-extension-option 4 paramspace english language en param max-time-vm-retry 2 param max-extension-length 4 param aa-pilot 3700 paramspace english location flash: param second-greeting-time 30 param welcome-prompt _bacd_welcome.au param queue-manager-debugs 1 param call-retry-timer 15 param max-time-call-retry 600 param voice-mail 3005 paramspace english prefix en param service-name queue ! ! ! BR2#dir flash: Directory of flash:/ 1 -rw- 24679 no date app-b-acd-2.1.0.0.tcl 2 -rw- 33870 no date app-b-acd-aa-2.1.0.0.tcl 3 -rw- 75650 no date en_bacd_allagentsbusy.au 4 -rw- 83291 no date en_bacd_disconnect.au 5 -rw- 63055 no date en_bacd_enter_dest.au 6 -rw- 37952 no date en_bacd_invalidoption.au 7 -rw- 496521 no date en_bacd_music_on_hold.au 8 -rw- 123446 no date en_bacd_options_menu.au 9 -rw- 42978 no date en_bacd_welcome.au 10 -rw- 496521 Mar 012002 01:13:09 +00:01 music-on-hold_3db.au 11 -rw- 496521 Mar 012002 02:47:07 +00:01 music-on-hold.au 536870908 bytes total (534895700 bytes free) BR2# BR2# There is no output when I do debug voice application script OUTPUT OF 'debug voice application session' is as below. Calling no: 3002, called no: 3700 BR2#debug voice app BR2#debug voice application sess voip application session debugging is on BR2# Mar 1 03:17:40: //37//AFW_:/Closing_AnyEvent: Mar 1 03:17:40: //37//AFW_:/Session_Cleaner: Mar 1 03:17:40: //-1//AFW_:/C_ServiceSession_Event_Handler: Mar 1 03:17:40: //37/8A066DCF802F/AFW_:/C_ServiceSession_Event_Handler:Received event CC_EV_CALL_DISCONNECT_DONE[17] in Main Loop Mar 1 03:17:40: //37//AFW_:/AFW_M_Session_Terminate: Mar 1 03:17:40: //-1//AFW_:HN000FF5F4:/AFW_M_Session_Free: MOD[Session_65BFE164_0_1046004]( ) Mar 1 03:17:42: //-1//AFW_
Re: [OSL | CCIE_Voice] BACD issue - No welcome prompt
Its local call from CME phone to bacd. No gatekeeper in between. PH1---CME with AA/ACD---ephone=hunt CC: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] BACD issue - No welcome prompt Date: Sat, 4 Oct 2008 11:36:07 -0700 Do a debug RAS and check if you see an ARJ from the gatekeeper. Try unregistering the CME from the GK and try. Any call that uses a VOIP dialpeer will require bandwidth authorization for 128kbps and if you have a bandwidth cac restriction within the cme zone the call will fail. Vik Malhi – CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] Join our free online support and peer group communities: http://www.IPexpert.com/communitiesIPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. On Oct 4, 2008, at 10:00 AM, Kapil Atrish wrote:The attached file has full config and debug output if you wish to see. ! dial-peer voice 15 voip destination-pattern 3700 session target ipv4:172.22.102.1 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 16 voip service aa incoming called-number 3700 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! Thanks for your time Subject: RE: [OSL | CCIE_Voice] BACD issue - No welcome prompt Date: Sat, 4 Oct 2008 12:39:38 -0400 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com Can you send your dial-peer for the BACDapplication? Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | [EMAIL PROTECTED]: 301.313.2041 | C: 443.789.1219 | www.presidio.com From: [EMAIL PROTECTED]:[EMAIL PROTECTED] On Behalf Of Kapil Atrish Sent: Saturday, October 04, 20087:58 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] BACDissue - No welcome prompt HI, Attached is my config. I get fast busy tone and Unknown number on display whenI dial the pilot number from any CME phone. I can dial hunt-pilot directly andcall get routed correctly or give the aa-pilot to hunt-pilot and ring thephones fine. Call in between phones are setup using G711ulaw. I've tried singlevoip dial-peer with incoming called-address and destination-pattern, reload ofrouter, re-configure script. Below is the snapshot of bacd config and debug voice application seesion.. application service queue flash:app-b-acd-2.1.0.0.tcl param queue-len 15 param aa-hunt5 3701 param queue-manager-debugs 1 param number-of-hunt-grps 2 ! service aa flash:app-b-acd-aa-2.1.0.0.tcl paramspace english index 1 param number-of-hunt-grps 2 param menu-timeout 6 param handoff-string aa param dial-by-extension-option 4 paramspace english language en param max-time-vm-retry 2 param max-extension-length 4 param aa-pilot 3700 paramspace english location flash: param second-greeting-time 30 param welcome-prompt _bacd_welcome.au param queue-manager-debugs 1 param call-retry-timer 15 param max-time-call-retry 600 param voice-mail 3005 paramspace english prefix en param service-name queue ! ! ! BR2#dir flash: Directory of flash:/ 1 -rw- 24679 no date app-b-acd-2.1.0.0.tcl 2 -rw- 33870 no date app-b-acd-aa-2.1.0.0.tcl 3 -rw- 75650 no date en_bacd_allagentsbusy.au 4 -rw- 83291 no date en_bacd_disconnect.au 5 -rw- 63055 no date en_bacd_enter_dest.au 6 -rw- 37952 no date en_bacd_invalidoption.au 7 -rw- 496521 no date en_bacd_music_on_hold.au 8 -rw- 123446 no date en_bacd_options_menu.au 9 -rw- 42978 no date en_bacd_welcome.au 10 -rw- 496521 Mar 012002 01:13:09 +00:01 music-on-hold_3db.au 11 -rw- 496521 Mar 012002 02:47:07 +00:01 music-on-hold.au 536870908 bytes total (534895700 bytes free) BR2# BR2# There is no output when I do debug voice application script OUTPUT OF 'debug voice application session' is as below. Calling no: 3002, called no: 3700 BR2#debug voice app BR2#debug voice application sess voip application session debugging is on BR2# Mar 1 03:17:40: //37//AFW_:/Closing_AnyEvent: Mar 1 03:17:40: //37//AFW_:/Session_Cleaner: Mar 1 03:17:40: //-1//AFW_:/C_ServiceSession_Event_Handler: Mar 1 03:17:40: //37/8A066DCF802F/AFW_:/C_ServiceSession_Event_Handler:Received event CC_EV_CALL_DISCONNECT_DONE[17] in Main Loop Mar 1 03:17:40: //37//AFW_:/AFW_M_Session_Terminate: Mar 1 03:17:40: //-1//AFW_:HN000FF5F4:/AFW_M_Session_Free: MOD[Session_65BFE164_0_1046004]( ) Mar 1
[OSL | CCIE_Voice] Interviewing CCIE candidates
i came to know that candidates are being interviewed by the proctor before sitting for the lab exam. Has anyone come across this and can share a bit about the i/v? _ Searching for the best deals on travel? Visit MSN Travel. http://in.msn.com/coxandkings
[OSL | CCIE_Voice] CSS - Partitions for later use
Hi, I was working on worbook volume 3 lab 2 and realised few partitions needs to be created and added to respective CSS at later stage, its about addig couple of partitions and updating 3 to 6 CSS. I thought of creating few dummy partitions in the begining itself and add to all the CSS. Couple before internal partition (which may be used for IPMA etc.) and couple after internal partition. For example, I create: 1) pt_br1_dummybeforePTinternal1 2) pt_br1_dummybeforePTinternal2 3) pt_hq_dummybeforePTinternal1 4) pt_hq_dummybeforePTinternal2 Create few more 5) pt_br1_dummyAfterPTinternal1 6) pt_br1_dummyAfterPTinternal2 7 pt_hq_dummyAfterPTinternal1 8 pt_hq_dummyAfterPTinternal2 Put these partitions in all BR1 and HQ CSS before and after pt_internal. Now, whenever I need to add a partitions, I simply rename one of them. Even if I leave any partition ununsed I don't see any harm but not sure how will that be graded. If anybody has followed this approach and any idea how it'll be graded? Thanks, Kapil Atrish _ Want to explore the world? Visit MSN Travel for the best deals. http://in.msn.com/coxandkings
[OSL | CCIE_Voice] XNTP/install.bat missing - NTP not installed on CCM
Hi, Today I was configuring NTP on CCM. I didn't find NTP service available in services console. I tried to install but install.bat was not present inside XNTP folder. I checked another CCM installation and the file was not there as well. Any thoughts how can i restore install.bat if missing from XNTP folder? Thanks for your time. _ Movies, sports news! Get your daily entertainment fix, only on live.com http://www.live.com/?scope=videoform=MICOAL
Re: [OSL | CCIE_Voice] IPPA URL
Hi, Following is the URL available in Lab exam now: http://www.cisco.com/web/psa/products/index.html Go to Voice Unified communications Cisco Unified CCM Maintain and Operate Guides there you'll find features and services guide... Jacob Owen [EMAIL PROTECTED] wrote: It's been awhile since i sat for the lab and I can't remember if they had removed the ability to use the UniverCD so I was wondering, if you can't use it now where do find the URL's for the various IPT services like IPPA, Extension Mobility, Fast Dials, and Personal Directory. If I am mistaken that we can't use the UniverCD please let me know. -- Jacob Owen CCIE #14063 (RS, Service Provider), CCDP, CCVP
Re: [OSL | CCIE_Voice] IPPA URL
For IPPA URL: PRODUCT SUPPORT VOICE AND UNIFIED COMMUNICATIONS CISCO AGENT DESKTOP INSTALL AND UPGRADE Install and Upgrade Guides See CAD installation guide. kapil atrish [EMAIL PROTECTED] wrote: Hi, Following is the URL available in Lab exam now: http://www.cisco.com/web/psa/products/index.html Go to Voice Unified communications Cisco Unified CCM Maintain and Operate Guides there you'll find features and services guide... Jacob Owen [EMAIL PROTECTED] wrote: It's been awhile since i sat for the lab and I can't remember if they had removed the ability to use the UniverCD so I was wondering, if you can't use it now where do find the URL's for the various IPT services like IPPA, Extension Mobility, Fast Dials, and Personal Directory. If I am mistaken that we can't use the UniverCD please let me know. -- Jacob Owen CCIE #14063 (RS, Service Provider), CCDP, CCVP
Re: [OSL | CCIE_Voice] Antw: Re: IPPA URL
Hi, This link doesn't even give access to SRND. Not sure, let's wait for someone who attends the lab to confirm. I got that URL info from cisco.com/go/certsupport. I raised a case specifially to know the new URL and they gave me this one. Thanks, Kapil Atrish Robert Schuknecht [EMAIL PROTECTED] wrote: Hi, i thought the new Link, for documentation, is more restrictive and it lets you only go to: http://www.cisco.com/web/psa/products/tsd_products_support_configure.html And today (24th September) should be the first day to use the new documentation, in the LAB. Does anyone knows which link is the right one? /Robert kapil atrish schrieb am Mittwoch, 24. September 2008 um 18:49 in Nachricht 94b3025f7ed9ac364355a477ec047e47: Hi, Following is the URL available in Lab exam now: http://www.cisco.com/web/psa/products/index.html Go to Voice Unified communications Cisco Unified CCM Maintain and Operate Guides there you'll find features and services guide... Jacob Owen wrote: It's been awhile since i sat for the lab and I can't remember if they had removed the ability to use the UniverCD so I was wondering, if you can't use it now where do find the URL's for the various IPT services like IPPA, Extension Mobility, Fast Dials, and Personal Directory. If I am mistaken that we can't use the UniverCD please let me know.
[OSL | CCIE_Voice] FRTS with PPP
I was working on eBook Volume 1 lab 4. Q 31. What I understood from there is that When doing PPP multilink fragmentation alogwith LLQ and frame-relay adaptive traffic-shaping, there is no class map to be associatecd to DLCI? Sample configuration: policy-map LLQ ! Policy-map GTS shape adaptive/average x serivce-policy LLQ ! interface virtual-template x ppp multilink commands... service-policy output GTS ! frame-relay interface dlci xxx ppp virtual-template x No Frame-relay class required I checked other examples also: If doing FRF.12 with LLQ and GTS or If doing FRF.12 with LLQ and FRTS or PPP Multilink fragmentation with FRTS and LLQ - frame-relay map class is required to be associated to DLCI. Is it correct to conclude that PPP fragmentation with adaptive-shaping is the only scenario where no class map to be associated to DLCI directly. Since adaptive-shaping needs to know about the traffic in LLQ, it should've the policy map LLQ statement. And Interface Virtual-template is going to do the fragmentation/interleaving hence apply the final Policy-Map to this interface. Looking forward for clarification on my understanding Thanks for your time _ Want to explore the world? Visit MSN Travel for the best deals. http://in.msn.com/coxandkings
[OSL | CCIE_Voice] Unity question
Hi, 1) Is there any way in Unity to check the detailed call treatment other then looking at the Port Status Monitor and Call Viewer? I want to check which Route Handler (Direct Call or Forwarded Call) the call hit initially and step by step flow from there onwards, if any option is available which can give detailed analysis? I've observed Port Status monitor/Call viewer don't tell exact rule name in certian scenarios. 2) Is there a way out I can create two mailboxes in Unity with same extension. I tried adding a new Dialing domain but no success. Perhaps someone has done it pl comment. Thanks for your time.. Voice your opinion on the burning issues of the day. Discuss, debate with the world. Logon to message boards on MSN. Try it! Hottest news and in-depth analysis that goes beyond the headlines. Only on MSN News Check it out! _ Search for videos of Bollywood, Hollywood, Mollywood and every other wood, only on Live.com http://www.live.com/?scope=videoform=MICOAL
[OSL | CCIE_Voice] Fax question
Hi, When Fax-Relay is enabled on 6608 module, what should be the Fax speed i.e. 7200bps or 14400 bps? Should it be configured 7200bps if far end is a Router and 1400bps if far end is a Cat 6608 module? What should it be set to if far end is an ATA/VG248? When fax-passthrough is configured on ATA, what should be the Audio mode set to i.e 0015 (Vad enabled) or 0014 (Vad disabled)? May it be cleary mentioned in lab whether Vad is required or not? Thanks for your time. _ Movies, sports news! Get your daily entertainment fix, only on live.com http://www.live.com/?scope=videoform=MICOAL
[OSL | CCIE_Voice] Not able to use Ring Subscriber at this extention in Unity
Hi, I've a subscriber (2001) configured in Unity. I want to route the call as following: If someone tries to reach 2001 and the call goes unanswered, it should ring at 2002 and if that extention also doesn't answer the call, the caller should get routed to Voice Mailbox of 2001. I configured a subscriber 2001 in Unity. On Call Transfer settings configured : Ring subscriber at this extention: 2002 (Supervisory Transfer). After 3 rings Unity should pull the call back. Under greetings I've configured unity to send the call to Voice mailbox of subscriber 2001. Ext 2001 is configured to route Forward No Answer to VM under CCM. My understanding of this configuration is that the call will first ring 2001, will get routed to Unity and ring 2002. If 2002 also doesn't answer, it'll get routed to VM of 2001. But with above configuration I am not able to transfer the call to 2002. I've achieved this by using a Call Handeler which is configured to ring subscriber at extention: 2002 and afterwards transfer to VM of 20001. Pl comment what am I doing wrong in first scenario. Thanks, _ Search for videos of Bollywood, Hollywood, Mollywood and every other wood, only on Live.com http://www.live.com/?scope=videoform=MICOAL
[OSL | CCIE_Voice] Need help in answering following queries
Hi, I've come up with few questions during my preparation. Require help in answering following queries: 1) What default COS-DSCP-MAP needs to be changed when configuring Campus wide QOS? I think only COS 5 needs to be mapped to 46-47 instead of 42-47 even if it is not asked in the Lab. set qos cos-dscp-map 0 10 18 26 34 46 48 56 Is there any change required in dscp to cos map as well? set qos dscp-cos-map 0-7:0 8-15:1 16-23:2 24-31:3 32-39:4 40-47:5 48-55:6 56-63:7 2) UniverCD is getting discontinued. Does someone know exactly what Cisco documentation (URL) would be available in place of UniverCD and what'll be the restrictions to access the same? 3) If codec sampling rate needs to be changed on H.323 GW (for ex for G.729 from 20ms to 30 ms), should we change only at GW level and CCM will negotiate or both sides need to be changed? How about MGCP Gateway? 4) Do we need to enable spanning-tree portfast on Cat6k port connecting to IP Phones? Say Lab exam doesn't mention but about enabling spanning-tree portfast but it does ask to configure data and voice vlan on IP Phone ports. 5) When using 6608-T1 or E1 with CAS, how can we configure it to use only first three time-slots? Is this done the same way as done for fractional PRI by configuring service parameters or by configuring only 1-3 ports for incoming/outgoing/both directions under MGCP gateway config? 6) When 6608-T1/E1 is configured for CAS, does the lab mention which type of signaling needs to be configured per timeslot for ex. wink-start,delay-start etc. or leave it default to wink-start? 7) In case of H.323 GW, if default channel-selection-order needs to be changed from bottom up to top down, is this need to be changed under isdn interface only or at CCM Gateway confiugration page as well? 8) While registering CCM Pub and Sub to GK with Tech prefix, GK does the load-balancing of calls to both the CCMs when using 1# as default-technology prefix. This behaviour can be overriden by configuring zone prefix test 1# 2... gw-priority 10 ccm_1 command. Now, if the previous command is not put in, what is the quickest and efficient way to check which CCM server the call was routed to i.e. Pub or Sub, to ensure GK is sending calls to both CCM Server and doing load-balancing? Is debug gatekeeper main 5/main 10 or some other wayout? 9) 1005 is a hunt-pilot having 1001 (HQ Phone 1) and 2001 (BR1 Phone 1). When a VM is left for 1005 during out of office hours all the phones should get MWI. Workbook solution (Create a CTIRP,forward to VM during out of office hours etc is done) creates a Public Distribution List and adds 1001 and 2001 to that list. Adds a Call Handeler for 1005 and points it to PDL created earlier to achieve this. Can't we create a mailbox for 1005 and configure alternate extention 1001 and 2001 also enable MWI for 1001 and 2001? Is this not a working solution? P 1- (126 IPExpert voice proctor guide) Thanks. Kapil Atrish Voice your opinion on the burning issues of the day. Discuss, debate with the world. Logon to message boards on MSN. Try it! _ Movies, sports news! Get your daily entertainment fix, only on live.com http://www.live.com/?scope=videoform=MICOAL
