[OSL | CCIE_Voice] %CDP-4-re: DUPLEX_MISMATCH: duplex mismatch discovered on HQ3750 SW...can not browse into CUCM (Mike Thompson)
There is a bug in the CUCM CDP module. I have seen it in production network and end up with disabling the CDP daemon onCUCM network services. There is a tech note on CCO. And a bug fix in next release. hth frog On Thu, Apr 8, 2010 at 3:21 PM, ccie_voice-requ...@onlinestudylist.comwrote: Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: %CDP-4-DUPLEX_MISMATCH: duplexmismatchdiscovered on HQ3750 SW...can not browse into CUCM (Mike Thompson) 2. Re: %CDP-4-DUPLEX_MISMATCH: duplexmismatchd -- Smile, you'll save someone else's day! Frog ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] sip issues
have you tried dtmf-relay rtp-nte ? -- Smile, you'll save someone else's day! Frog
[OSL | CCIE_Voice] sip issues
also voip service voice sip to sip h323 to sip blah blah ? -- Smile, you'll save someone else's day! Frog
Re: [OSL | CCIE_Voice] vpim
On Mon, Dec 22, 2008 at 1:56 PM, FrogOnDSCP46EF ciscoboy2...@gmail.comwrote: Vpim in 5 minutes: http://www.voiceie.com/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=8;t=001915 -- Smile, you'll save someone else's day! Frog -- Smile, you'll save someone else's day! Frog
[OSL | CCIE_Voice] how to sync two CCM6x clusters USER database wihout AD integration
'Hi Guys, Here is the requirement. 1. There are 2 clusters - Cluster1 and cluster2 2. users created in cluster1 should replicate to cluster2 automatically and vice-versa. 3. No 3rd party e.g. MS-AD integration , just 2 ccm servers v6/7x As far as my research goes, I can do above manually using BAT tool. Export from cluster1 and import in cluster2 and voice-versa. IS there any method to automate this? -- Smile, you'll save someone else's day! Frog
[OSL | CCIE_Voice] CUE Voicemail + VPIM in 10 minutes
Check the blog below! pushkarbhatkoti.wordpress.com http://pushkarbhatkoti.wordpress.com/2008/08/15/cue-voicemail-vpim-networking-cue-to-unity-in-10-minutes/ -- Smile, you'll save someone else's day! Frog
[OSL | CCIE_Voice] QoS any scenario in 10 minutes only
Is it possible? http://pushkarbhatkoti.wordpress.com/2008/08/07/a-practical-approach-qos-in-10-minutes/ -- Smile, you'll save someone else's day! Frog
Re: [OSL | CCIE_Voice] A practical approach- QoS in 10 minutes (any scenario)
Ben NG on ask expert forum has confirmed that yes, its allowed as long as your solution satisfies the question's requirement! http://forums.cisco.com/eforum/servlet/NetProf?page=netprofforum=Expert%20Archivetopic=Career%20CertificationsCommCmd=MB%3Fcmd%3Dpass_through%26location%3Doutline%40 ^1%40%40.eeac675 hehe try on real router, mine always worked. I just tried again and it worked! -frog On Fri, Aug 8, 2008 at 12:26 AM, Jonathan Charles [EMAIL PROTECTED] wrote: Ok, I just tried it (on a 3745 in GNS3) and it did not create the virtual template... Jonathan On Thu, Aug 7, 2008 at 8:40 AM, Jonathan Charles [EMAIL PROTECTED]wrote: Sure, it is possible... the question is, will they 0 out your points for using auto-qos, my guess is yes. Also, since you can't do ccm config why would you be allowed to do auto qos? Or am I insane to think that auto qos is not allowed? We have seen serious issues with auto-qos (in the real world... it doesn't work as well as one would hope...) Jonathan On Thu, Aug 7, 2008 at 6:03 AM, FrogOnDSCP46EF [EMAIL PROTECTED]wrote: is it possible? how? http://pushkarbhatkoti.wordpress.com/2008/08/07/a-practical-approach-qos-in-10-minutes/ -- Smile, you'll save someone else's day! Frog -- Smile, you'll save someone else's day! Frog
[OSL | CCIE_Voice] The only CCIE voice certified FROG on this planet - yeah I finally passed it
The only frog on this planet with CCIE# Not joking, frog went 4th time to the CCIE lab in Sydney and this time asked to the proctor;- hey, I have been to this lab for 3 times and u never been kind enough to gave me the #. This time he got serious and thought a while, then he tabbed frog's head and put his hand in his pocket and gave frog a brand new CCIE#. Frog hopped in pound again! who's going to use his CCIE#? no its damn wastage ! Seriously if someone want to hire frog in Sydney , PM him. I finally got my shiny new number after studying hard for the past 9 months with my full time job. In Oct, 2005 I started studying for RS lab but after a few month of preparation I though its not enough, end of the day its vendor cert. I decided to leave CCIE for a while and gain a Masters Degree in Networking first. I took an admission in Charles Stuart University Sydney and then finished it in a year a month [jan 2006 to feb 2007]. Then I had nothing to do so I thought to go back and do CCIE cert. I took a few months break from study and then thought seriously about which CCIE track I go for. I finally made a decision do the toughest exam first and then easier ones, so I hopped on CCIE voice horse. At the beginning I started studying 2-6 hours per week for CCIE voice written after june2007. I passed the voice written exam in august 2007. After that it was a real challenge for me to arrange routers switches, servers and softwares for the home lab. It took 2 months to arrange routers, servers for unity, call manager etc. I started labbing from nov, 2007 and binded up my study on 20th of July,2008 [ in 9 months]. I have logged about 1700+ hours of labbing and studying. 80% labbing and 20% studying on the forums, workbook, cisco.com/unvercd etc. To be honest, I didn't sleep properly for the last 4 months. I used to get back to home form work at 6pm, start labbing until 3-4am. If I remember correctly I didn't sleep for more than 3-4 hours per day, after that get up early morning7am, get ready and go to work. I found it worked for me but the drawback of this is - in middle of the week I used to collapse completely due to not enough sleep. but end of the day, hard work pays! Thanks to everyone who've helped me. Special thanks to my boss, and my employer for supporting me - yes they paid for my 4 attempts and about $30K for my lab gears [lucky me]. Oh yeah, proctor at Sydney lab is very helpful [that doesn't mean that he gives you the solutions]. I have read many threads on GS or other forums saying that proctor changes config during lunch break or something like that. Hey, the fact is that the proctor goes with you to have his lunch as well so who is there to change the configs? all hoax! Also, in Sydney lab proctor were very helpful in provide the DocCD docs [the link which were not working]. Thanks Scott for your help and I won't forgot those nice eggs sandwiches. Now I will take a few weeks break before I start RS track. Here is the list of my study materials; This forum [voiceie.com] was very helpful and was the main tool [along with cisco.com/univercd] for my study. IM study mates; this is a must tool for everyone, make your MSN/yahoo study partner. This helps when u run into trouble and your available resources gets short. U can instant ask the question to your study mates. I had 2-3 good full time 24/7 study mates. without them I couldn't have done it. FYI, I never attended any bootcamps or institute for VoIP training. I am working in IT for the past 8 years [mainly in RS, firwalls, iptel stuffs] so it was easy for me to nail all topics of voice exam. I have used a MOC lab from Robert Hockley [kiwi guy] http://www.ccievoice-assessor.com/ that was helpful and I came across knowing many thing which I wouldn't have normally picked up by myself. its was about $450 bucks but after I did my 8 hours full lab, we [robert and I] went through all scenarios, questions and discussed on the phone for more than 5 hours. WoW u can ask him anything virtually.. Apart from above, I used ccbootcamp's volume#1 and technology workbook. the technology workbook was helpful when i was starting to study. so for new aspirant its good to use ccbootcamp's technology workbook. Thanks Brad and Avner for such a wonderful quick reference book. COD – Fasial khans and Mark snow/Vik's free online class. Take free online from these experts, its worth. Specially Vik's IPMA tricks [ a must see]. How to do ipma using single partion. Thanks VIK for that, really helpful. I also used IEmentor's voice workbook as a reference and to simulate different scenario. Also bought a few books on call manager, gw/gatekeeper but found them helpless. If you are thinking to buy one of those books, i would rather advise to buy ccbootcamp's technology book. Those books still on my self and badly need dusting. [image: [Razz]] ) Remember, you've to come up with many scenarios as possible. The last
[OSL | CCIE_Voice] Conf max sessions
hi there, I think u only have pvdn2-8 in ur router. U can not do conference with it. the reason; 1. u probably have a T1 interface - which steals depends on u rtime slot configured 1-3 on e1/t1 interface, that many DSP from pvdm2-8 2. now lets say u have configure 3 timeslot - 8-3 = 5 timeslot remaining for other stuffs xocing / conferencing 3. xcode should work withoiut any issue bcoz u have 5 session remaining 4. Conf-bridge takes the whole DSP chip a.k.a u need to put another pvdm2-8 and do conferecing on that. u can not share pvdm2-8 for everything. hth -- Smile, you'll save someone else's day! Frog From: Onur Tufekci [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] 0 Conf max sessions To: [EMAIL PROTECTED] Cc: ccievoice1 [EMAIL PROTECTED], OSL CCIE Voice Lab Exam ccie_voice@onlinestudylist.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=windows-1252 Here is the configuration and results: router2(config)#do show run Building configuration... Current configuration : 2938 bytes ! ! Last configuration change at 15:11:09 UTC Mon Jul 7 2008 ! NVRAM config last updated at 14:49:04 UTC Mon Jul 7 2008 ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec service password-encryption ! hostname router2 ! boot-start-marker boot-end-marker ! card type t1 0 0 no logging console enable secret 5 $1$7cJ4$U9gWuPCv.H1DPdXYIyWJD0 ! no aaa new-model ! resource policy ! network-clock-participate wic 0 ! ! ip cef ! ! no ip domain lookup ! isdn switch-type primary-ni ! voice-card 0 dspfarm dsp services dspfarm ! ! ! ! ! ! ! ! ! ! ! !
Re: [OSL | CCIE_Voice] Conf max sessions
Exactly..Cheers On Tue, Jul 8, 2008 at 1:42 AM, Onur Tufekci [EMAIL PROTECTED] wrote: Thank you bunch for the explanation. I was reading about it and trying to figure out how it works. You are right I have 1 DSP looks like. NAME: PVDMII DSP SIMM with one DSP on Slot 0 SubSlot 4, DESCR: PVDMII DSP SIMM with one DSP PID: PVDM2-16 , VID: V01 , SN: FOC1032054R On Mon, Jul 7, 2008 at 11:34 AM, FrogOnDSCP46EF [EMAIL PROTECTED] wrote: hi there, I think u only have pvdn2-8 in ur router. U can not do conference with it. the reason; 1. u probably have a T1 interface - which steals depends on u rtime slot configured 1-3 on e1/t1 interface, that many DSP from pvdm2-8 2. now lets say u have configure 3 timeslot - 8-3 = 5 timeslot remaining for other stuffs xocing / conferencing 3. xcode should work withoiut any issue bcoz u have 5 session remaining 4. Conf-bridge takes the whole DSP chip a.k.a u need to put another pvdm2-8 and do conferecing on that. u can not share pvdm2-8 for everything. hth -- Smile, you'll save someone else's day! Frog From: Onur Tufekci [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] 0 Conf max sessions To: [EMAIL PROTECTED] Cc: ccievoice1 [EMAIL PROTECTED], OSL CCIE Voice Lab Exam ccie_voice@onlinestudylist.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=windows-1252 Here is the configuration and results: router2(config)#do show run Building configuration... Current configuration : 2938 bytes ! ! Last configuration change at 15:11:09 UTC Mon Jul 7 2008 ! NVRAM config last updated at 14:49:04 UTC Mon Jul 7 2008 ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec service password-encryption ! hostname router2 ! boot-start-marker boot-end-marker ! card type t1 0 0 no logging console enable secret 5 $1$7cJ4$U9gWuPCv.H1DPdXYIyWJD0 ! no aaa new-model ! resource policy ! network-clock-participate wic 0 ! ! ip cef ! ! no ip domain lookup ! isdn switch-type primary-ni ! voice-card 0 dspfarm dsp services dspfarm ! ! ! ! ! ! ! ! ! ! ! ! -- Smile, you'll save someone else's day! Frog
[OSL | CCIE_Voice] GDM Log-in
Hi Onue, It sounds like your situation is the same as Jack Skellingtonhttp://en.wikipedia.org/wiki/Jack_Skellingtonin The nightmare before Christmas ( I was shocked to see that 3D version is available now) I think your quesiton is already answered and is a valid one :) -frog -strip- Onur. Yes, there is no need to create separate voice mail box for 2nd line, just use the same mailbox to access the GDM, the phone belongs to one person only. There is no way that you could access GDM directly by pressing 9 without logging in to the mailbox first. What u need to do is put 2nd lines DN as E.164 under first lines DN settings in CUE. After this when u take line 2 and press message key then it will ask for password only and then once u logged in you can press 9 to access GDM. ---strip -- Smile, you'll save someone else's day! Frog Message: 2 Date: Mon, 7 Jul 2008 12:22:38 -0400 From: Onur Tufekci [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] GDM Log-in To: Juan [EMAIL PROTECTED] Cc: o Ninja [EMAIL PROTECTED], ccie_voice@onlinestudylist.com, [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=utf-8 Hi Vik, Mark, This subject is getting frustrated (I sure am) since no one is able to figure out what exactly is going on. Is it possible for you guys to shed some light on this? Best Regards, Onur. - Onur. Yes, there is no need to create separate voice mail box for 2nd line, just use the same mailbox to access the GDM, the phone belongs to one person only. There is no way that you could access GDM directly by pressing 9 without logging in to the mailbox first. What u need to do is put 2nd lines DN as E.164 under first lines DN settings in CUE. After this when u take line 2 and press message key then it will ask for password only and then once u logged in you can press 9 to access GDM.
[OSL | CCIE_Voice] bacd call ringing time adjustment
ephone-dn 10 pilot 5000 member 5001, 5002 timeout 5 condition 1# max-time-call-retry = 60 ( a must) condition 2# the hunt-group should seize by members twice only ( e.g. 5001 ring first, then 5002 again 5001 and then 5002, after that call shud drop within 60 seconds when they reach max-time-call-retry). I hvae done this question; max-time-call-retry 60 call-retry-timer 30 ( mind you' 30 is the maximum limit) but it still call hunts 5001 followed by 5002, repeat *5002 then 5002, then again 5001 only.. (2.5 times) The condition is the hunt-gropu should only seiezd for two time only. -- Smile, you'll save someone else's day! Frog
[OSL | CCIE_Voice] debug isdn q931 equivalent command in CCM
debug isdn q931 is a life saver, does anyone know how to find out what number ccm is getting from remote gw/phoens to troubleshoot the dialplan. SDL traces are way too complicated and I never worked them out where they show what numbenr CCM got and why it was rejected. May be i am looking the wrong log file. Thanks in advance. -- Smile, you'll save someone else's day! Frog
Re: [OSL | CCIE_Voice] pstn calls during AAR
Thanks Vik, I had the concept wrong and a friend of mine corrected me last night. I was thinking that call goes to ccm first and then comes back to siteb which got me confused. Cheers Message: 3 Date: Tue, 10 Jun 2008 22:40:20 -0700 From: Vik Malhi [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] pstn calls during AAR To: 'OSL CCIE Voice Lab Exam' ccie_voice@onlinestudylist.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii If a call destined for a San Jose phone comes into the San Jose gateway, why should the WAN being congested be a concern (for this particular call). So in other words, Locations CAC bandwidth deductions do not occur for calls within a Location (San Jose in the example). Locations CAC comes into play when you have a call into or out of the Location San Jose. Vik Malhi - CCIE #13890 Senior Technical Instructor - IPexpert, Inc. -- Smile, you'll save someone else's day! Frog
[OSL | CCIE_Voice] GDM e.164 number
I was testing the following scenario; Scenario#1; ephone-dn1 number 6001 ephone-dn 2 number ephone-dn3 number 6002 ephone-dn 4 number ephone-1 button 1:1 2:2 ephone 2 button 1:3 2:4 VM works okay for 6001 and 6002. I wanted to test from button 2 of each phone so that when user press button#2 of phone1or2 and press msg button they shud be prompted for by cisco cue lady saying Enter your password I simply added and in 6001 6002's E-164 number. It worked as expected. I press line 2 of first phone (dn#) and then pressed msg button and i was able to check 6001's mailbox after entering the password. Scenario#2; --- same DN as above... Then I wanted to test it for GDM voicemail. I created a GDM mailbox. CUE# groupname GDM1 create CUE# groupname phonenumber 1000 CUE# groupname phonenumberE164 --this dn is 2nd line on phone1, phone1username on cue is phone1 CUE# groupname GDM1 member phone1 phone1 is member of gdm1 CUE# voicemail mailbox owner GDM1 size 2000 CUE# voicemail callerid Now, I picked up the line#2 (dn) of phone1 and then pressed MSG button and CUE lady prompts me for enter your ID. Here I was expecting to enter only PASSWORD as it did for above scenario#1. then i tried xlation pattern to change the ANI of to 1000. but i can see CME was sending ANI=1000 to CUE but still can't access GDM mailbox using just a password. Anyone have tested it successfully? Frog -- Smile, you'll save someone else's day! Frog
[OSL | CCIE_Voice] ipexpert way to xlate the incoming outgoing ani dnis
I am quite furious to know that why ipexpert does the xlation pattern in below way. lets say call from pstn to hq site needs to be cut downto 4 digit so that phone with 4 digit DID can ring straight away... 4082022001 to 2001 in ipexpert's all workbook the method is being used: == rule 1 /^408...\(2...\)/ /\1/ while the same result can be achieved by using simply; rule 1 /^4082022/ /2/ Why make life complicated? no KISS*** concept? perhaps I am missing the point where i can't use **my KISS*** method and ipexpert method will kick in in that case! -- Smile, you'll save someone else's day! crazy! Frog
Re: [OSL | CCIE_Voice] ipexpert way to xlate the incoming
Hi Vik, Genius!, now I stood why you use that one. Thanks heaps for clarification. I got caught into the issue you've mentioned - TEHO calls to sitec from hq and I decided to do ani based on the per dial-peer. so create a teho dialpeer and that was sorting the things up for me. but now ipexpert method seems I will have to use only one command!, not on all individual dialpeers! Another solution could be, add another line in sitec; rule 1 /^2/ /6175272/ rule 2 /^21222/ /921222/ this one for teho? note; TEHO (ANI 2122211003) Will that work? i didn't test it. cheers frog Date: Fri, 6 Jun 2008 07:22:05 -0700 From: Vik Malhi [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] ipexpert way to xlate the incoming outgoinganidnis To: 'OSL CCIE Voice Lab Exam' ccie_voice@onlinestudylist.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii In the case you have given your translation woud be sufficient. We have gone with the number slice or \(\) method for a very good reason and this is more to do with expansion of digits. Think about the rule you would use to expand 2... to 6175272... (in the case of ANI in the outbound direction) What rule would you use in this case? With the easier way of doing this you would use: rule 1 /^2/ /6175272/ Now when HQ uses this site gw in the case of TEHO (ANI 2122211003) what would happen to this ANI? There would also be a match. The translated number would be 61752722122211003 which is no good. This is why the following rule would be a better solution. rule 1 /^\2...\)$/ /617521\1/ As a general rule- BE SPECIFIC TO AVOID OVERLAP. Vik Malhi - CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] Join our free online support and peer group communities: http://www.IPexpert.com/communities http://www.ipexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- Smile, you'll save someone else's day! Frog
[OSL | CCIE_Voice] pattern matching
Hi group, pattern#1 = destination-pattern 7[^4]...$ pattern#2 = destination-pattern 74...$ Can anyone tell me whats the difference between patter1 vs pattern 2? -- Smile, you'll save someone else's day! Frog
Re: [OSL | CCIE_Voice] pattern matching
Thanks Mark and matthew. This was really tricky. normally i use rule 1 ^515 the marker there without squire bracket matches string which start with 515 Cheers -- Smile, you'll save someone else's day! Frog Message: 2 Date: Wed, 4 Jun 2008 18:59:30 -0700 From: Matthew Bynum [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] pattern matching To: OSL CCIE Voice Lab Exam ccie_voice@onlinestudylist.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 The ^ imediately after the open bracket ( as in
[OSL | CCIE_Voice] Subject: BACD Queuing
Reload the router.I have had this issue too. -- Smile, you'll save someone else's day! Frog
[OSL | CCIE_Voice] CCM : location based CAC MOH bandwidth consideration
Folks, Allow 2 calls between HQ and SiteB , HQ to siteb are in G729 region. 2 calls = 24kbps per call x 2 = 48 kbps What about MOH bandwidth if HQ is pumping M.Cast MOH to SITEB? Think about when both calls are occupied and 48kbps bandwidth CAC is exhausted and we also want to put someone on MOH? How CCM maths work in that situation? Frog -- Smile, you'll save someone else's day! Frog
Re: [OSL | CCIE_Voice] CCM : location based CAC MOH bandwidthconsideration
Christian, I think one stream (MOH) should consume what is configured in the srevice parameter under mediaapp. The case I depicted here is G729 region (hq to siteB). So... Back to the my original question, bandwidth consideration, 2xg729 calls vs 3 x G729 calls. If you set 3 calls, (considering 1xg729 for MOH), you may get 0 in HA section. Frog On Wed, Apr 30, 2008 at 11:32 PM, Christian Narvaez [EMAIL PROTECTED] wrote: You can test it using Permon , and selecting the Performance Object Cisco Locations then see how the BandwidthAvailable varies. -Original Message- From: [EMAIL PROTECTED] on behalf of FrogOnDSCP46EF Sent: Wed 4/30/2008 5:45 AM To: CCIE Voice Maillist Subject: [OSL | CCIE_Voice] CCM : location based CAC MOH bandwidthconsideration Folks, Allow 2 calls between HQ and SiteB , HQ to siteb are in G729 region. 2 calls = 24kbps per call x 2 = 48 kbps What about MOH bandwidth if HQ is pumping M.Cast MOH to SITEB? Think about when both calls are occupied and 48kbps bandwidth CAC is exhausted and we also want to put someone on MOH? How CCM maths work in that situation? Frog -- Smile, you'll save someone else's day! Frog -- Smile, you'll save someone else's day! Frog
Re: [OSL | CCIE_Voice] ATA sccp version 3.xx auto registration with CCM
I Thanks Edward, regarding If your device is not registering make sure the TFTP is correct and the EID is . Do I neeed to define the TFTP address in ATA? I thot that just assigning EID . and CM0 cm1 field it will register automatic. As for TFTP asisgnment mannually to the ATA unit, my understanding is that ATA does dhcp broadcast, gets an ip address and DHCP pushes' TFTP setting to the ATA using DHCP OPTION 150. Anyhow, thanks for pointing it out, may be ATA doesn't have a code to accept the Option 150. I will test it and post my observation. Cheers Frog On Mon, Apr 28, 2008 at 12:05 PM, Edward French [EMAIL PROTECTED] wrote: Typically the ATA autoregisters no problem. As for the Mac address the first port will register with the device mac address and the second port will register with a modified address for example say your mac is 00AB12345678 this would be the mac for the first port (port 0) the second port would register as AB1234567801. If your device is not registering make sure the TFTP is correct and the EID is . Ed - Original Message From: FrogOnDSCP46EF [EMAIL PROTECTED] To: CCIE Voice Maillist ccie_voice@onlinestudylist.com Sent: Sunday, April 27, 2008 9:13:15 PM Subject: [OSL | CCIE_Voice] ATA sccp version 3.xx auto registration with CCM has anyone managed to get ATA-186 sccp version 3.x auto-registration with CCM? I had to add it manually. I restarted CCM services after setting the DHCP to 1. It gets IP address okay I can get access to the unit. Also I noticed that the MAC address in ATA configuration page is different than what it actually have. not sure why but its there. So be careful and just look for arp -a or show cdp neighbor MAC. -- Smile, you'll save someone else's day! Frog -- Smile, you'll save someone else's day! Frog
[OSL | CCIE_Voice] ATA sccp version 3.xx auto registration with CCM
has anyone managed to get ATA-186 sccp version 3.x auto-registration with CCM? I had to add it manually. I restarted CCM services after setting the DHCP to 1. It gets IP address okay I can get access to the unit. Also I noticed that the MAC address in ATA configuration page is different than what it actually have. not sure why but its there. So be careful and just look for arp -a or show cdp neighbor MAC. -- Smile, you'll save someone else's day! Frog
[OSL | CCIE_Voice] MLPPP doubt
Hi group, MLPPP can be achieved using: 1. virtual-template (single interface) 2. multilink PPP group (grouping interfaces) Can we use multilink PPP group for the exam purpose? (normally all workbook tells to configure virtual-template) -- Smile, you'll save someone else's day! Frog
Re: [OSL | CCIE_Voice] How to find quick port numbers in the exam forCat6k QoS
I couldn't behave myself and finally found the easiest way to find the port numbers quickly in the exam: Just remember the question's wording... e..g. h323 sip mgcp skinny Jump on your flashy Cisco IOS and issue these commands rack02r1#show ip port-map | in mgcp Default mapping: mgcp udp port 2427 system defined rack02r1# rack02r1#show ip port-map | in h323 Default mapping: h323 tcp port 1720 system defined Default mapping: h323callsigalt tcp port 11720 system defined Default mapping: h323callsigalt udp port 11720 system defined rack02r1# rack02r1#show ip port-map | in sip Default mapping: sip-tls tcp port 5061 system defined Default mapping: sip-tls udp port 5061 system defined Default mapping: sip udp port 5060 system defined rack02r1# rack02r1# isn't that easy ? Why to remember if there is so many other things to remember? Cheers Frog
Re: [OSL | CCIE_Voice] IPCC: unable to connect agent to the IPPA service
Hi Greg, yeah, both are associated - i.e. Jtapi and RMJTAPI, also Agent with phone number 5008 is associated with the ICD ext. Any other hint? *User : crs user* Status: Ready First NameLast Name*User ID jtapi_1 User Password*PIN * Telephone NumberManager User IDDepartmentUser LocaleNone English United States Enable CTI Application Use Enable CTI Super Provider Call Park Retrieval Allowed Enable Calling Party Number Modification Name Dialing Not Defined Associated PC Not Defined Primary Extension Not Defined ICD Extension Not Defined Controlled Devices CRS_4001, CRS_4002, CRS_4003, RP_CRS_4000 Enable Authentication Proxy Rights False Controlled Device Profiles none * indicates required item. View page inEnglish, United States -- Page displayed at Tue Apr 22 10:08:15 CDT 2008 Copyright (c) 1999 - 2004 Cisco Systems, Inc. All rights reserved RM user: - *User : crs user* Status: ReadyFirst NameLast Name*User ID rm User Password*PIN *Telephone NumberManager User IDDepartmentUser Locale None English United States Enable CTI Application Use Enable CTI Super Provider Call Park Retrieval Allowed Enable Calling Party Number Modification Name Dialing Not Defined Associated PC Not Defined Primary Extension Not Defined ICD Extension Not Defined Controlled Devices CRS_4001, CRS_4002, CRS_4003, RP_CRS_4000, SEP001D45432537 Enable Authentication Proxy Rights False Controlled Device Profiles none * indicates required item. View page inEnglish, United States On Tue, Apr 22, 2008 at 9:56 PM, Gregory Jost (grjost) [EMAIL PROTECTED] wrote: Did you also associate with rmjtapi user? Greg Jost Network Consulting Engineer Unified Communications Practice Cisco Systems, Inc. 214-274-1922 -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *FrogOnDSCP46EF *Sent:* Monday, April 21, 2008 7:44 PM *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] IPCC: unable to connect agent to the IPPA service Hi there, My IPPA service in CCM Pub is running when I try to login as a agent to IPCC it gives me an error Unable to connect to the IPPA service I have created an user on CCM and associated the HQ phone1(5001) to the user. I varified the user name and password as well using http://10.1.2.19/ccmuser/ and found it authenticates okay. I am not runing IPMA and EM on the pub so no chances of getting any port conflict. Any pointer? -- Smile, you'll save someone else's day! Frog -- Smile, you'll save someone else's day! Frog
[OSL | CCIE_Voice] How to find quick port numbers in the exam for Cat6k QoS
ANy idea? I found a url ACL based and but it doesn't have all ports for QoS. So just wondering if someone can can give some tips on quickly finding the port numbers to put in the cat6k QoS. here is the link.. http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/srnd/design/guide/endpts.html contains straight copy and paste no need to memorize. CAT6500 (enable) *set qos cos-dscp-map 0 8 16 24 32 46 48 56 * CAT6500 (enable) *set qos policed-dscp-map 0, 24, 46:8 * CAT6500 (enable) CAT6500 (enable) *set qos policer aggregate VVLAN-VOICE rate 128 burst 8000 drop * CAT6500 (enable) *set qos policer aggregate VVLAN-CALL-SIGNALING rate 32 burst 8000 policed-dscp * CAT6500 (enable) *set qos policer aggregate VVLAN-ANY rate 5000 burst 8000 policed-dscp * CAT6500 (enable) *set qos policer aggregate PC-DATA rate 5000 burst 8000 policed-dscp * CAT6500 (enable) CAT6500 (enable) *set qos acl ip IPPHONE-PC dscp 46 aggregate VVLAN-VOICE udp* *Voice_IP_Subnet Subnet_Mask* *any range 16384 32767 * CAT6500 (enable) *set qos acl ip IPPHONE-PC dscp 24 aggregate VVLAN-CALL-SIGNALING tcp* *Voice_IP_Subnet Subnet_Mask* *any range 2000 2002 * CAT6500 (enable) *set qos acl ip IPPHONE-PC dscp 0 aggregate VVLAN-ANY* *Voice_IP_Subnet Subnet_Mask* *any * CAT6500 (enable) *set qos acl ip IPPHONE-PC dscp 0 aggregate PC-DATA any * CAT6500 (enable) *commit qos acl IPPHONE-PC * CAT6500 (enable) *set vlan* *vvlan_id mod/port * CAT6500 (enable) *set port qos* *mod/port* *trust-device ciscoipphone * CAT6500 (enable) *set qos acl map IPPHONE-PC *mod/port CAT6500 (enable) -- Smile, you'll save someone else's day! Frog
Re: [OSL | CCIE_Voice] SRST call restriction labbook 8.9
did i get it right? yeah i also noticed that your css and partition have the same name, it is recommended approach? KISS*** Sara *FrogOnDSCP46EF [EMAIL PROTECTED]* wrote:
[OSL | CCIE_Voice] SRST call restriction labbook 8.9
Hi Sara, The example config you've pasted on the forum shows: condition1: DN:2002 should be able to dial Internal. condition2: DN:2002 should receive all incoming call too. (assuming CSS_ALL have everything, as u didn't paste full config). In a nutshell, here is what my understanding is about the COR. Lets say DN2002 should able to dial everyone in SRST but no PSTN caller should be able to dial 2002. step1: create partiotions dial-peer voice cor custom name 911 name local name ld name intl Step2: create cor list: dial-peer voice cor list 911 css same as in CCM member 911 --partition created in step1 dial-peer voice cor list local member local dial-peer voice cor list ld member ld dial-peer voice cor list intl member intl note: above 4 we will apply to the POTS dialpeer dial-peer voice 3 pots corlist outgoing 911 noticed the list above we created? destination-pattern 911 port 0/2/0:23 forward-digits 3 ! dial-peer voice 4 pots destination-pattern 9[2-9].. corlist outgoing local port 0/2/0:23 forward-digits 7 ! dial-peer voice 5 pots destination-pattern 91[2-9]..[2-9].. corlist outgoing ld port 0/2/0:23 forward-digits 11 ! dial-peer voice 10 pots destination-pattern 9011T corlist outgoing outgoing ld port 0/2/0:23 prefix 011 ! Now we've made a pitch for our COR so far... now time to look at the question and prepare for it step3: Create custom CORLIST now for restricting phone dn 2002 (incoming restricted but outgoing it shud be able to dial everone. *Outgoing calls from 2002:* dial-peer voice corlist DN2002 member 911 ---will be able to access the dial-peer POTS 911, local, ld , intl (call to everyone from 2002 dn) member local member ld member intl Apply above corlist in SRST configuration: config t call-manager-fallback corlist outgoing DN2002 1 2002-- CORLIST DN2002 is applied to phone DN 2002. thats all, 2002 will be able to call everyone... now work on PSTN callier calling DN 2002 in SRST.*. (INCOMING to 2002)* We can achieve this task in 2 way... My prefereed way just create a translation profile and apply that to srst-fallback: voice translation-rule 1 rule 1 reject /^2002/ voice translation-profile BLOCK2002incoming translate called 1 config t call-manager-fallback translation-profile incoming BLOCK2002incoming or voice-port 0/2/0:23 voice translation-profile incoming block2002incoming thats all.. in the lab book 8.9 there is a requirement to configure cor so that no pstn caller to call 2002 in srst mode. the config given is: cor incoming css-911 1 2002-2003 cor incoming css-all 2 2002 cor outgoing css-intl 3 2002 dial-peer voice 2 pots corlist incoming css-911 incoming called number . direct-inward-dial dial-peer css-intl member pt-internal dial-peer css-911 member pt-911 -- Smile, you'll save someone else's day! Frog
Re: [OSL | CCIE_Voice] Multiple ANI DNIS in Unity Call routing
Hi Christian, I understand that but my original question was: how can I type 300* and 800* in unity call routing's ANI/DNIS field. Frog.. On Wed, Apr 16, 2008 at 12:11 AM, Christian Narvaez [EMAIL PROTECTED] wrote: You've got to use for instance : The first rule with: 300* (block from 3001 to 3009) and The Second rule with 800* (block from 8001 to 8009) -Original Message- From: [EMAIL PROTECTED] on behalf of FrogOnDSCP46EF Sent: Mon 4/14/2008 10:16 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Multiple ANI DNIS in Unity Call routing If I had to put multiple block of ANI/DNIS in Call routing (forwaring) in Unity how can I do it? Normally it will only let me put either single ANI/DNIS or wildmask. e..g 3002 or 300X but I want to put 300x + 800X - Smile, you'll save someone else's day! Frog -- Smile, you'll save someone else's day! Frog
[OSL | CCIE_Voice] Multiple ANI DNIS in Unity Call routing
If I had to put multiple block of ANI/DNIS in Call routing (forwaring) in Unity how can I do it? Normally it will only let me put either single ANI/DNIS or wildmask. e..g 3002 or 300X but I want to put 300x + 800X - Smile, you'll save someone else's day! Frog
Re: [OSL | CCIE_Voice] Test B-ACD script w/o pstn connection
just one more question, i didnt hear the welcome message of the in welcom_prompt, i double checked the au file on my flash matches the param setting. -- Sara, You have done good job. Just make sure your prompt file should be like this: show flash: en_bacd_welcome.au and in Bacd config it should be like this: param welcome-prompt _bacd_welcome.au U will hear the prompt if u strip out the en. Cheers - Frog
Re: [OSL | CCIE_Voice] Test B-ACD script w/o pstn connection
Exactly, welcome file comes with the script plays only thank you for calling thats all. Just tried playing that welcome file in my media player and it also says thanks for calling May be u playing wrong file. Cheers Frog On Sun, Apr 13, 2008 at 6:12 PM, [EMAIL PROTECTED] wrote: thanks Frog, it is working now...i noticed that the file it played is thank you for calling ...but if i play the au file on my media player, it plays welcome is that supposed to be like that? cheers! Sara *FrogOnDSCP46EF [EMAIL PROTECTED]* wrote: just one more question, i didnt hear the welcome message of the in welcom_prompt, i double checked the au file on my flash matches the param setting. -- Sara, You have done good job. Just make sure your prompt file should be like this: show flash: en_bacd_welcome.au and in Bacd config it should be like this: param welcome-prompt _bacd_welcome.au U will hear the prompt if u strip out the en. Cheers - Frog -- GANBARE! NIPPON! Win your ticket to Olympic Games 2008.http://pr.mail.yahoo.co.jp/ganbare-nippon/ -- Smile, you'll save someone else's day! Frog
Re: [OSL | CCIE_Voice] MGCP and SRST
I was just wondering how this tracking business will be done in SRST. Assume AAR connection is up between siteB HQ and suddenly the serial link goes down. Frog --- I think somehow the status of the serial interface is linked to the PRI link by some connection ID or something because when the serial interface is down, that connection ID is lost. When you bring the serial interface backup again, the serial interface must generate a new ID or something and it cannot associate that to the PRI link.
Re: [OSL | CCIE_Voice] Dialing HQ phones to CME BACD-AA number
Hi Johathan, Thats already enabled. Xcoder is also inplace. Still no go... Call shows its connected on the IP phone screen but silence and after 30 second it gets disconnected. THe debug voip dialpeers doesn't tell anything. On Mon, Apr 7, 2008 at 12:56 PM, Jonathan Charles [EMAIL PROTECTED] wrote: Enable an IPIPGW voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip By default all of the above are disabled. Jonathan On Sun, Apr 6, 2008 at 7:32 PM, FrogOnDSCP46EF [EMAIL PROTECTED] wrote: Anybody has working solution for this scenario? HQ phones to CME BACD AA won't work. while it works from the PSTN phone. similar issue: http://www.voiceie.com/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=8;t=000944 -- Smile, you'll save someone else's day! Frog -- Smile, you'll save someone else's day! Frog
[OSL | CCIE_Voice] Dialing HQ phones to CME BACD-AA number
Anybody has working solution for this scenario? HQ phones to CME BACD AA won't work. while it works from the PSTN phone. similar issue: http://www.voiceie.com/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=8;t=000944 -- Smile, you'll save someone else's day! Frog
[OSL | CCIE_Voice] Gatekeeper
RAS ip is optional not necessary. I was just wondering if it wasn't there what will happen? Will it choose random interface for GK discovery? Frog *ras-IP-address* (Optional) IP address of one of the interfaces on the gatekeeper. When the gatekeeper responds to gatekeeper discovery messages, it signals the endpoint or gateway to use this address in future communications. *Note *Setting this address for one local zone makes it the address used for all local zones. -- Smile, you'll save someone else's day! Frog
