Re: [OSL | CCIE_Voice] Cisco ripped me off
all, yesterday i took my second attempt in rtp, and i am 200 % sure that i pass the exam. I got 1 hour left even after testing it thrice, but looking at the score report i was shocked, and i completely disagree with my score report. F... CCIE lab script evaluation.. i am completely pissed off the way it showed the results... no more CCIE in my life... i appreciate all my friends who helped me in this journey. thank you krishna. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Cisco ripped me off
Cory, Technically speaking, the grading has to be evaluated by taking the seating position where we took the exam rather doing it remotely for their convenience. i used switchport mode trunk, switchport trunk native vlan data on sb and sc. Can anyone expect fail in the exam after evaluating the tasks thrice and check everything line by line, and the end showing the score report as fail... This is completely insane. I was wondering if i can legally proceed so that justification will be done for the right candidates. Thank you krishna. From: Cory Gray corygray22...@hotmail.com To: 'Krishna' vinayak_...@yahoo.com; 'Online Study' ccie_voice@onlinestudylist.com Sent: Wednesday, October 31, 2012 7:41 AM Subject: RE: [OSL | CCIE_Voice] Cisco ripped me off Krishna, I am sorry to hear that. I suffered something similar during my last attempt but after much thinking I think I know what happened and maybe the same happened to you. Even though IPexpert recommends using switchport mode trunk on ESW interfaces I still had been using switch mode access because it never failed. I also did this because using switchport mode trunk would show nothing in the show vlan-switch command so I was scared this was how it was being graded and would miss the points. IPexpert recommends this because they say the other way has been known to stop working for no reason. When I got my score report the next day, I could see several sections wrong that I knew I configured right. Doing the math I believe when they went to grade my exam the next day that my CUCME phones were no longer registered. I will use switchport mode trunk for now on. What did you do? That is my only theory. Maybe you have one different that can help others if you choose not to take it again. I will be back 11/30 and am hoping to do as well as I did last time but pass J Thanks, Cory From:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Krishna Sent: Wednesday, October 31, 2012 8:08 AM To: Online Study Subject: Re: [OSL | CCIE_Voice] Cisco ripped me off all, yesterday i took my second attempt in rtp, and i am 200 % sure that i pass the exam. I got 1 hour left even after testing it thrice, but looking at the score report i was shocked, and i completely disagree with my score report. F... CCIE lab script evaluation.. i am completely pissed off the way it showed the results... no more CCIE in my life... i appreciate all my friends who helped me in this journey. thank you krishna. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Cisco ripped me off
Hassan, regarding gatekeeper also i tested everything line by line, and i am pretty sure everything is working as expected and also regarding domain name for gatekeeper i approached proctor and he says use whatever you want for gatekeeper call routing between hq,sb and sc. I shutdown the gatekeeper twice and check everything and works as expected. I think CCIE is not written in my fate, and so its better to leave this journey rather holding it for none. thank you krishna. From: Mohamed Hassan mrmha...@gmail.com To: Krishna vinayak_...@yahoo.com Cc: Online Study ccie_voice@onlinestudylist.com Sent: Wednesday, October 31, 2012 8:06 AM Subject: Re: [OSL | CCIE_Voice] Cisco ripped me off you remind me of my last attempt, i was coming our from exam and confident that i will pass, but i found the score full of zeros :((( but maybe one question like gatekeeper or QoS causes to fail in all questions. One more attempt friend and let's do it. On Wed, Oct 31, 2012 at 3:08 PM, Krishna vinayak_...@yahoo.com wrote: all, yesterday i took my second attempt in rtp, and i am 200 % sure that i pass the exam. I got 1 hour left even after testing it thrice, but looking at the score report i was shocked, and i completely disagree with my score report. F... CCIE lab script evaluation.. i am completely pissed off the way it showed the results... no more CCIE in my life... i appreciate all my friends who helped me in this journey. thank youkrishna. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- Engineer / Mohamed Rabea Unified communication engineer___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Cisco ripped me off
Cory, i really appreciated your motivation skills, but looking at the score report i am unable to understand where i did wrong and so how can i comprehend myself by looking at the lab result, and taking exam one more time. i would say that cisco should give us a statement feedback for what had caused 0 points for that many tasks, otherwise we would never come to know what we did wrong. thank you krishna. From: Cory Gray corygray22...@hotmail.com To: 'Krishna' vinayak_...@yahoo.com; 'Online Study' ccie_voice@onlinestudylist.com Sent: Wednesday, October 31, 2012 9:22 AM Subject: RE: [OSL | CCIE_Voice] Cisco ripped me off Krishna, I had some funny things going on with my rack but cannot get into it because of NDA. I am extremely frustrated. I would tell you this. I am already a CCIE in RS and work for Cisco. I am not sure if you have CCIE yet. I know it is frustrating, expensive, and time consuming but as you can see from the last few weeks, several people on this list have passed. Me and you are so far along (I was done an hour early also) that the worst thing you can do is give up now. All of the effort you put in to get this far will be wasted if you do not complete your journey. It took me a few days to get over it. Get back in there as soon as possible and knock it out! Especially if you do not have any CCIE’s, passing this exam will be a career defining moment that will help you more than any project or customer experience you can think of. Don’t quit! From:Krishna [mailto:vinayak_...@yahoo.com] Sent: Wednesday, October 31, 2012 10:14 AM To: Cory Gray; 'Online Study' Subject: Re: [OSL | CCIE_Voice] Cisco ripped me off Cory, Technically speaking, the grading has to be evaluated by taking the seating position where we took the exam rather doing it remotely for their convenience. i used switchport mode trunk, switchport trunk native vlan data on sb and sc. Can anyone expect fail in the exam after evaluating the tasks thrice and check everything line by line, and the end showing the score report as fail... This is completely insane. I was wondering if i can legally proceed so that justification will be done for the right candidates. Thank you krishna. From:Cory Gray corygray22...@hotmail.com To: 'Krishna' vinayak_...@yahoo.com; 'Online Study' ccie_voice@onlinestudylist.com Sent: Wednesday, October 31, 2012 7:41 AM Subject: RE: [OSL | CCIE_Voice] Cisco ripped me off Krishna, I am sorry to hear that. I suffered something similar during my last attempt but after much thinking I think I know what happened and maybe the same happened to you. Even though IPexpert recommends using switchport mode trunk on ESW interfaces I still had been using switch mode access because it never failed. I also did this because using switchport mode trunk would show nothing in the show vlan-switch command so I was scared this was how it was being graded and would miss the points. IPexpert recommends this because they say the other way has been known to stop working for no reason. When I got my score report the next day, I could see several sections wrong that I knew I configured right. Doing the math I believe when they went to grade my exam the next day that my CUCME phones were no longer registered. I will use switchport mode trunk for now on. What did you do? That is my only theory. Maybe you have one different that can help others if you choose not to take it again. I will be back 11/30 and am hoping to do as well as I did last time but pass J Thanks, Cory From:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Krishna Sent: Wednesday, October 31, 2012 8:08 AM To: Online Study Subject: Re: [OSL | CCIE_Voice] Cisco ripped me off all, yesterday i took my second attempt in rtp, and i am 200 % sure that i pass the exam. I got 1 hour left even after testing it thrice, but looking at the score report i was shocked, and i completely disagree with my score report. F... CCIE lab script evaluation.. i am completely pissed off the way it showed the results... no more CCIE in my life... i appreciate all my friends who helped me in this journey. thank you krishna. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Cisco ripped me off
Marko, i opened case with support, and i hope i will get some good feedback . I appreciate your help. thank you krishna. From: Marko Milivojevic mar...@ipexpert.com To: Krishna vinayak_...@yahoo.com Cc: Cory Gray corygray22...@hotmail.com; Online Study ccie_voice@onlinestudylist.com Sent: Wednesday, October 31, 2012 1:13 PM Subject: Re: [OSL | CCIE_Voice] Cisco ripped me off Krishna, You can always open a case with the certification support and state your reasons. While there is no regrade for the voice lab, Cisco has in the past issued retake vouchers for people who have been wronged by the grading... if it was indeed the grading issue. -- Marko Milivojevic - CCIE #18427 (SP RS) Senior CCIE Instructor - IPexpert On Wed, Oct 31, 2012 at 7:13 AM, Krishna vinayak_...@yahoo.com wrote: Cory, Technically speaking, the grading has to be evaluated by taking the seating position where we took the exam rather doing it remotely for their convenience. i used switchport mode trunk, switchport trunk native vlan data on sb and sc. Can anyone expect fail in the exam after evaluating the tasks thrice and check everything line by line, and the end showing the score report as fail... This is completely insane. I was wondering if i can legally proceed so that justification will be done for the right candidates. Thank you krishna. From: Cory Gray corygray22...@hotmail.com To: 'Krishna' vinayak_...@yahoo.com; 'Online Study' ccie_voice@onlinestudylist.com Sent: Wednesday, October 31, 2012 7:41 AM Subject: RE: [OSL | CCIE_Voice] Cisco ripped me off Krishna, I am sorry to hear that. I suffered something similar during my last attempt but after much thinking I think I know what happened and maybe the same happened to you. Even though IPexpert recommends using switchport mode trunk on ESW interfaces I still had been using switch mode access because it never failed. I also did this because using switchport mode trunk would show nothing in the show vlan-switch command so I was scared this was how it was being graded and would miss the points. IPexpert recommends this because they say the other way has been known to stop working for no reason. When I got my score report the next day, I could see several sections wrong that I knew I configured right. Doing the math I believe when they went to grade my exam the next day that my CUCME phones were no longer registered. I will use switchport mode trunk for now on. What did you do? That is my only theory. Maybe you have one different that can help others if you choose not to take it again. I will be back 11/30 and am hoping to do as well as I did last time but pass J Thanks, Cory From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Krishna Sent: Wednesday, October 31, 2012 8:08 AM To: Online Study Subject: Re: [OSL | CCIE_Voice] Cisco ripped me off all, yesterday i took my second attempt in rtp, and i am 200 % sure that i pass the exam. I got 1 hour left even after testing it thrice, but looking at the score report i was shocked, and i completely disagree with my score report. F... CCIE lab script evaluation.. i am completely pissed off the way it showed the results... no more CCIE in my life... i appreciate all my friends who helped me in this journey. thank you krishna. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Cisco ripped me off
All, I have a very strong evidence that supports my statement, and in case if i don't get good feedback or positive response, I will approach lawyer and take it to the court for justice so that no others would get affect in future. thank you krishna. From: Leslie Meade leslie.me...@lvs1.com To: vinayak_...@yahoo.com vinayak_...@yahoo.com; mar...@ipexpert.com mar...@ipexpert.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Wednesday, October 31, 2012 2:06 PM Subject: Re: [OSL | CCIE_Voice] Cisco ripped me off I seem to remember that someone has been down this path and it did not end well. They touted the we do not do remarks blah blah... But I hope you get a better response Sent from Samsung Mobile Krishna vinayak_...@yahoo.com wrote: Marko, i opened case with support, and i hope i will get some good feedback . I appreciate your help. thank you krishna. From: Marko Milivojevic mar...@ipexpert.com To: Krishna vinayak_...@yahoo.com Cc: Cory Gray corygray22...@hotmail.com; Online Study ccie_voice@onlinestudylist.com Sent: Wednesday, October 31, 2012 1:13 PM Subject: Re: [OSL | CCIE_Voice] Cisco ripped me off Krishna, You can always open a case with the certification support and state your reasons. While there is no regrade for the voice lab, Cisco has in the past issued retake vouchers for people who have been wronged by the grading... if it was indeed the grading issue. -- Marko Milivojevic - CCIE #18427 (SP RS) Senior CCIE Instructor - IPexpert On Wed, Oct 31, 2012 at 7:13 AM, Krishna vinayak_...@yahoo.commailto:vinayak_...@yahoo.com wrote: Cory, Technically speaking, the grading has to be evaluated by taking the seating position where we took the exam rather doing it remotely for their convenience. i used switchport mode trunk, switchport trunk native vlan data on sb and sc. Can anyone expect fail in the exam after evaluating the tasks thrice and check everything line by line, and the end showing the score report as fail... This is completely insane. I was wondering if i can legally proceed so that justification will be done for the right candidates. Thank you krishna. From: Cory Gray corygray22...@hotmail.commailto:corygray22...@hotmail.com To: 'Krishna' vinayak_...@yahoo.commailto:vinayak_...@yahoo.com; 'Online Study' ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Sent: Wednesday, October 31, 2012 7:41 AM Subject: RE: [OSL | CCIE_Voice] Cisco ripped me off Krishna, I am sorry to hear that. I suffered something similar during my last attempt but after much thinking I think I know what happened and maybe the same happened to you. Even though IPexpert recommends using switchport mode trunk on ESW interfaces I still had been using switch mode access because it never failed. I also did this because using switchport mode trunk would show nothing in the show vlan-switch command so I was scared this was how it was being graded and would miss the points. IPexpert recommends this because they say the other way has been known to stop working for no reason. When I got my score report the next day, I could see several sections wrong that I knew I configured right. Doing the math I believe when they went to grade my exam the next day that my CUCME phones were no longer registered. I will use switchport mode trunk for now on. What did you do? That is my only theory. Maybe you have one different that can help others if you choose not to take it again. I will be back 11/30 and am hoping to do as well as I did last time but pass J Thanks, Cory From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Krishna Sent: Wednesday, October 31, 2012 8:08 AM To: Online Study Subject: Re: [OSL | CCIE_Voice] Cisco ripped me off all, yesterday i took my second attempt in rtp, and i am 200 % sure that i pass the exam. I got 1 hour left even after testing it thrice, but looking at the score report i was shocked, and i completely disagree with my score report. F... CCIE lab script evaluation.. i am completely pissed off the way it showed the results... no more CCIE in my life... i appreciate all my friends who helped me in this journey. thank you krishna. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.platinumplacement.com/___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out
Re: [OSL | CCIE_Voice] access-list not matching any packets for mgcp ports on switch
i created an acl that calls mgcp ports i.e. udp 2427 2428 with extended acl permit tcp any any eq 2428, permit tcp any eq 2428 any , permit udp any any eq 2427, permit udp any eq 2427 any. i called the acl in the class map, where the class map is referenced in the policy map with appropriate bandwidth and qos configuration. i applied acl on the trunk port that connects to router. when i issued show access-lists, i am not seeing any matches on the acl and so i was wondering how could i verify that whether i am doing it right way or not.. any help is much appreciated. thank you krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] access-list not matching any packets for mgcp ports on switch
ip access-list extended 101 permit tcp any any eq 2428 permit udp any any eq2427 permit tcp any eq 2428 any permit udp any eq 2427 any class-map match-any c-mgcp match access-group name 101 policy-map p-mgcp class c-mgcp set dscp cs3 police 64000 8000 exceed-action drop int fa 1/0/1 --- trunk port to router mls qos trust dscp service-policy input p-mgcp From: Cory Gray corygray22...@hotmail.com To: Kevin Spicer ke...@kevinspicer.co.uk Cc: Krishna vinayak_...@yahoo.com; Online Study ccie_voice@onlinestudylist.com Sent: Wednesday, October 24, 2012 12:32 PM Subject: Re: [OSL | CCIE_Voice] access-list not matching any packets for mgcp ports on switch If you paste your config, we can be of better help Sent from my iPhone On Oct 24, 2012, at 12:23 PM, Kevin Spicer ke...@kevinspicer.co.uk wrote: This is on 3750 switch? Did you enable qos globally? (mls qos) On 24 Oct 2012 17:03, Krishna vinayak_...@yahoo.com wrote: i created an acl that calls mgcp ports i.e. udp 2427 2428 with extended acl permit tcp any any eq 2428, permit tcp any eq 2428 any , permit udp any any eq 2427, permit udp any eq 2427 any. i called the acl in the class map, where the class map is referenced in the policy map with appropriate bandwidth and qos configuration. i applied acl on the trunk port that connects to router. when i issued show access-lists, i am not seeing any matches on the acl and so i was wondering how could i verify that whether i am doing it right way or not.. any help is much appreciated. thank you krishna. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] cme ip add or loopback ip in CUE??
yes cory you're correct about understanding the question exactly. thank you krishna. From: Cory Gray corygray22...@hotmail.com To: Rrcrumm rrcr...@yahoo.com Cc: Krishna vinayak_...@yahoo.com; Online Study ccie_voice@onlinestudylist.com Sent: Saturday, October 20, 2012 8:23 PM Subject: Re: [OSL | CCIE_Voice] cme ip add or loopback ip in CUE?? Randall, I believe the question is when going through the GUI initialization and it ask for the IP address of CME, what IP address do you use? Krishna, Please correct me if I am wrong. Sent from my iPhone On Oct 20, 2012, at 9:06 PM, Rrcrumm rrcr...@yahoo.com wrote: The labs say to use an IP address if 10.10.115.2. So under the lo1 you need to add ip unnumbered lo1(also make sure to add the OSPG statement if needed and clear the ip OSPG process$ Then add the IP address and default gateway. Then make sure to add the static route Hth Randall On Oct 20, 2012, at 2:25 PM, Krishna vinayak_...@yahoo.com wrote: when using loopback address for CUE setup, does it matter whether what ip address we put it for cme in cue .i.e. for example loopback 10.10.115.1 or 10.10.202.1(cme ip address)... it works in both the cases but just want to make sure which one is the right way of doing it.. thank you krishna. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] cme ip add or loopback ip in CUE??
Dan Cory, thanks for your feedback.. i thought the same but i was triggered by looking at the ipexpert solutions guide for 5 lab handbook and so taking second thought from expertise guys like you.. thank you krishna. From: Dan Quinlan (daquinla) daqui...@cisco.com To: Cory Gray corygray22...@hotmail.com Cc: Online Study ccie_voice@onlinestudylist.com Sent: Saturday, October 20, 2012 9:10 PM Subject: Re: [OSL | CCIE_Voice] cme ip add or loopback ip in CUE?? Cory, I read the question the same way you did. Krishna- either will work if you're not using strict match in telephony service; however, I'd always match whatever address you use in telephony service. Chances are the lab is going to specify what address cme phones should use - I'd stay consistent with that. DQ d...@cisco.com On Oct 20, 2012, at 9:23 PM, Cory Gray corygray22...@hotmail.com wrote: Randall, I believe the question is when going through the GUI initialization and it ask for the IP address of CME, what IP address do you use? Krishna, Please correct me if I am wrong. Sent from my iPhone On Oct 20, 2012, at 9:06 PM, Rrcrumm rrcr...@yahoo.com wrote: The labs say to use an IP address if 10.10.115.2. So under the lo1 you need to add ip unnumbered lo1(also make sure to add the OSPG statement if needed and clear the ip OSPG process$ Then add the IP address and default gateway. Then make sure to add the static route Hth Randall On Oct 20, 2012, at 2:25 PM, Krishna vinayak_...@yahoo.com wrote: when using loopback address for CUE setup, does it matter whether what ip address we put it for cme in cue .i.e. for example loopback 10.10.115.1 or 10.10.202.1(cme ip address)... it works in both the cases but just want to make sure which one is the right way of doing it.. thank you krishna. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] LAN Qos questions
steffen, your approach is not right way of doing it because when u look the threshold values of the queues you have allocated max threshold is 100 and reserved threshold is 100, guess what both threshold i.e. t1 and t2 takes up to 100% value when desired and that being said after t1 and t2 were filled it comes to t3 which has 75% i.e. it is the last threshold where it will take/borrow the memory value from reserved threshold when desired. long story short... right way of doing it either assign it to t2 or t1 and assign threshold value of 75% for correct approach... thank you krishna. From: Steffen Bruening stbruen...@gmail.com To: Pixar Perfect pixarperf...@live.com Cc: ccie_voice@onlinestudylist.com Sent: Saturday, October 20, 2012 6:38 PM Subject: Re: [OSL | CCIE_Voice] LAN Qos questions I have this seen this also, to be honest I think it shouldn't matter whether it is in threshold 1 or 3 as long as no other COS is in same Threshold of queue 1 of queset 2. When you leave in in threshold 3 I think you should be fine with: mls qos queue-set output 2 threshold 1 100 10075 100. Maybe I am completly wrong but thats they way I understood this. Regards Steffen 2012/10/20 Pixar Perfect pixarperf...@live.com The requirement is as follows on Lab 2 QOS section of the 5-Lab Handbook. For traffic being sent to the Site A gateway ensure that the traffic marked with COS 5 is dropped if the queue 1 is 75% full The Solution guide (page 408) has the following solution. mls qos queue-set output 2 threshold 1 75 100 100 100 -- queset is preconfigured on the port to 2 mls qos srr-queue output cos-map queue 1 threshold 3 5 .. My interpretation was to move the Cos 5 into Q1t1 but the command says threshold 3 .. is this just a typo or am I missing something obvious. Thanks! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Meet Me question - CME GW
all, added to this question it needs tones for callers who join and who leave the conference.. i configure voice class custom-cptones, assigned frequency and cadency but still cannot hear any join or leave tones.. i also assigned this cptones to dspfarm profile as well... any idea what else it needs to make this work.. thank you krishna. From: Bruno Nonogaki brun...@gmail.com To: Kevin Spicer ke...@kevinspicer.co.uk Cc: Online Study ccie_voice@onlinestudylist.com; Krishna vinayak_...@yahoo.com Sent: Wednesday, October 17, 2012 4:05 PM Subject: Re: [OSL | CCIE_Voice] Meet Me question - CME GW Kevin, This approach sounds good. I will try tomorrow and let you know! Thanks, Bruno On Wed, Oct 17, 2012 at 5:32 PM, Kevin Spicer ke...@kevinspicer.co.uk wrote: Another slightly different idea... Instead of a dummy ephone-dn how about using a voip dial-peer... Voice Serv voip Allow h to h Dial-p v 4321 voip Destination-p 4321 Incoming called-number 4321 Session target ipv4:x.x.x.x Code g711u On 17 Oct 2012 20:33, Bruno Nonogaki brun...@gmail.com wrote: Krishna, By doing this, ephones that doesn't have after-hours exempt cannot join the conference. And neither the PSTN Phone. I am trying to work with cor lists, but no success: dial-peer cor custom name meetme dial-peer cor list to-meetme member meetme dial-peer cor list block-meetme ephone-dn 10 octo number 4321 corlist outgoing to-meetme conference meetme ephone-dn 1 octo ! -- no access to meet-me number corlist incoming block-meetme ephone-dn 2 octo ! -- access to meet-me, no corlist applied number Ephone-dn 1 can initiate meetme and PSTN phones can join it. Ephone-dn 2 cannot initiate meetme, but it is unable to join. I tried to setup another ephone-dn with a forward all to meet-me number, but it doesn't work. Very tricky question... if you find the answer, please post it! Regards, Bruno On Wed, Oct 17, 2012 at 4:22 PM, Krishna vinayak_...@yahoo.com wrote: Randell, under telephony-service define after-hours-block pattern , after-hours day mon 0:00 24:00, repeat it for days tue,wed,thurs,friday,sat,sunday. under ephone-dn where the phones can access meetme number apply after-hours-exempt... and i m sure you will be good. thank youkrishna. From: Rrcrumm rrcr...@yahoo.com To: William Bell b...@ucguerrilla.com Cc: Online Study ccie_voice@onlinestudylist.com Sent: Wednesday, October 17, 2012 2:03 PM Subject: Re: [OSL | CCIE_Voice] Meet Me question - CME GW Hi One requirement is all phones need to have a meet me button Thx Rc On Oct 17, 2012, at 11:34 AM, William Bell b...@ucguerrilla.com wrote: This may be an oversimplified approach, but couldn't you remove the MeetMe softkey from the ephone via an ephone-template? -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Oct 17, 2012, at 1:54 PM, Randall Crumm wrote: Hello, I am working on a CME Meet-me question(See below) My question is how do I restrict SCPH2 from iniatiating the MEET ME conference? Question: SITE CPh1 can initiate the meet me conference The other users can call the meet me number and get connected to the conference. PSTN can also access the conference bridge. - 4321 is the number for the meet me. - Make sure when user join and leave the conference beeps are heard - Only SC Ph1 should have only access to initiate the Meetme numbers, even though Softkeys should be available for all the users. - Only SCph1 should be able to see the conference participants. My config: ephone-dn 5 octo-line number 4321 no-reg primary conference meetme ! ephone-template 1 conference drop-mode local softkeys idle Redial Newcall Cfwdall Pickup Dnd softkeys seized Pickup Cfwdall Endcall Redial Meetme softkeys connected Hold Endcall Trnsfer Confrn Park ! ephone 1 ephone-template 1 ! ephone 2 ephone-template 1 ! voice class custom-cptone leave dualtone conference frequency 300 cadence 300 250 ! voice class custom-cptone Join dualtone conference frequency 700 cadence 300 50 sccp local GigabitEthernet0/1.102 sccp ccm 142.1.66.254 identifier 1 version 7.0 sccp ! sccp ccm group 1 bind interface GigabitEthernet0/1.102 associate ccm 1 priority 1 associate profile 2 register sc-cfb ! dspfarm profile 2 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 3 conference-join custom-cptone Join conference-leave custom-cptone leave associate application SCCP ! telephony-service sdspfarm units 2 sdspfarm tag 2 sc-cfb conference hardware max-conferences 12 gain -6 Have a great day! Thanks, Randall ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Meet Me question - CME GW
Randell, under telephony-service define after-hours-block pattern , after-hours day mon 0:00 24:00, repeat it for days tue,wed,thurs,friday,sat,sunday. under ephone-dn where the phones can access meetme number apply after-hours-exempt... and i m sure you will be good. thank you krishna. From: Rrcrumm rrcr...@yahoo.com To: William Bell b...@ucguerrilla.com Cc: Online Study ccie_voice@onlinestudylist.com Sent: Wednesday, October 17, 2012 2:03 PM Subject: Re: [OSL | CCIE_Voice] Meet Me question - CME GW Hi One requirement is all phones need to have a meet me button Thx Rc On Oct 17, 2012, at 11:34 AM, William Bell b...@ucguerrilla.com wrote: This may be an oversimplified approach, but couldn't you remove the MeetMe softkey from the ephone via an ephone-template? -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Oct 17, 2012, at 1:54 PM, Randall Crumm wrote: Hello, I am working on a CME Meet-me question(See below) My question is how do I restrict SCPH2 from iniatiating the MEET ME conference? Question: SITE CPh1 can initiate the meet me conference The other users can call the meet me number and get connected to the conference. PSTN can also access the conference bridge. - 4321 is the number for the meet me. - Make sure when user join and leave the conference beeps are heard - Only SC Ph1 should have only access to initiate the Meetme numbers, even though Softkeys should be available for all the users. - Only SCph1 should be able to see the conference participants. My config: ephone-dn 5 octo-line number 4321 no-reg primary conference meetme ! ephone-template 1 conference drop-mode local softkeys idle Redial Newcall Cfwdall Pickup Dnd softkeys seized Pickup Cfwdall Endcall Redial Meetme softkeys connected Hold Endcall Trnsfer Confrn Park ! ephone 1 ephone-template 1 ! ephone 2 ephone-template 1 ! voice class custom-cptone leave dualtone conference frequency 300 cadence 300 250 ! voice class custom-cptone Join dualtone conference frequency 700 cadence 300 50 sccp local GigabitEthernet0/1.102 sccp ccm 142.1.66.254 identifier 1 version 7.0 sccp ! sccp ccm group 1 bind interface GigabitEthernet0/1.102 associate ccm 1 priority 1 associate profile 2 register sc-cfb ! dspfarm profile 2 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 3 conference-join custom-cptone Join conference-leave custom-cptone leave associate application SCCP ! telephony-service sdspfarm units 2 sdspfarm tag 2 sc-cfb conference hardware max-conferences 12 gain -6 Have a great day! Thanks, Randall ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] lan qos- cos 5 threshold 1 in queue 1 doesn't show in running config
hi guys, i was wondering whether i am doing right way of doing lan qos or not ?? the requirements are assign cos 5 to priority queue , cos 3 4 to queue 2 with 60% exceed of cos 4 should be dropped. so here is my configuration for that mls qos mls qos srr-queue output cos-map queue 1 threshold 1 5 mls qos srr-queue output cos-map queue 2 threshold 2 3 mls qos srr-queue output cos-map queue 2 threshold 1 4 mls qos queue-set output 2 threshold 3 60 100 100 272 when i issued show run | i mls commands, i see every mls qos command except the cos 5 which is assigned to q1 t1. Is my approach is correct in dealing this question correctly?? does it matter whether we assign cos values to t1 or t2 or t3 in the queues??? your input is much appreciated. thank you krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] cme as srst bug??
hi guys, for h323 site during srst(cme with auto provision none), i couldn't able to restrict the max-calls-per-button and busy-trigger-per-button using srst ephone template.. and moreover until i put the hunstop channel command the call didn't go to voicemail though busy-trigger-per button is enabled with 1 . Is this is a bug for cme as srst ??? any help is much appreciated. thank you krishna. here is the config: SiteB-RTR(config)#do sh run | s ephone srst ephone template 1 max-ephones 10 ephone-dn-template 1 call-forward busy 2220 call-forward noan 2220 timeout 20 huntstop channel 1 ephone-template 1 softkeys remote-in-use Newcall CBarge max-calls-per-button 2 busy-trigger-per-button 1 SiteB-RTR(config)#do sh run | s telep telephony-service sdspfarm units 2 sdspfarm tag 1 SB-CFB srst mode auto-provision none srst ephone template 1 srst dn template 1 srst dn line-mode octo max-ephones 10 max-dn 10 no-reg ip source-address 10.10.65.254 port 2000 timeouts interdigit 3 system message fallback mode time-zone 8 time-format 24 voicemail 2220 max-conferences 8 gain -6 call-forward pattern .T transfer-system full-consult transfer-pattern .T ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] MGCP failover events
Amaia, follow the blog.ipexpert.com for right information on mgcp debug events... This blog is incorrect in showing the right output for pstn phone is ringing.. thank you krishna. From: Amaia Lesta amaia.le...@gmail.com To: ccie_voice@onlinestudylist.com Sent: Thursday, September 27, 2012 11:28 AM Subject: Re: [OSL | CCIE_Voice] MGCP failover events Hi It might be too late :( Chek the following post. It contains all the answers and explanations for this question http://dreamforccie.wordpress.com/2010/08/07/understanding-mgcp-packets-a-brief-overview-and-example-with-debugs/ BR Amaia ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CCIE Lab Strategy
Raffel, Don't expect good result on your first attempt. I am sorry if i am harsh, but the reality is Cisco will never let the student to pass right away in the first lab attempt and this would be exception to a few persons lets say 1 out 200 who pass the exam for the very first time. I took the exam first time and failed, and I swear its not that easy. you can imagine that I didn't had time to configure all the tasks for the lab. I just completely trusted my route plan and went ahead and implemented, and also just want to let you know the questions are not very very very clear. I am not repent for failing the exam since i felt this is not a big deal for the following reasons: 1.) the questions are not clear 2.) first time takers they give you every task that touches every device i.e. all technologies by the time when a student makes multiple attempts the lab will be little easy, and so grading as well. this fact is covered in kevin wallace video. I don't work for CISCO and therefore i have to pay from the pocket, and apparently i would say don't be repent if the results are not in your favor. once again i am stressing the wording of the questions and the exact requirements is very very very unclear After looking at my score report, i remember kevin's wallace statement about his first lab attempt... just let me know if you have any questions... thank you krishna. From: Raffel Enderson endersonraf...@yahoo.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Tuesday, September 25, 2012 2:19 PM Subject: [OSL | CCIE_Voice] CCIE Lab Strategy Hi, Great forum. Good work guys. I am so unlucky that i found this forum so late. Guys i am going for my first attempt this friday. Need strategies, i am very new to this forum and can any one help me with CCIE VOICE strategies. I am in desperate need of one. I have heard from CCIE's that we need to make our own strategies but i would like to know strategies who have already passed this lab. I m in desperate need. Thanks in advance. Rgds, Raffel ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] WAN QoS questions
1.) it depends, for example if this is a router with switch module in it i.e. BR1, i would not use trust since the marking will be done with acls by auto qos voip 2.) it depends, if you're told to use class based compression then you have to use compression header ip rtp in the policy-map otherwise you can leave as in the interface-dlci 3.) its once again you've to clarify with proctor.. for a full T1 i have never 95% bandwidth so far ... the bandwidth for =768 i have seen using 95% of the bandwidth speed. From: Randall Crumm rrcr...@yahoo.com To: Online Study ccie_voice@onlinestudylist.com Sent: Tuesday, September 18, 2012 11:30 AM Subject: [OSL | CCIE_Voice] WAN QoS questions Hello experts, I have some questions on WAN QoS. 1. If not explicitly not told to TRUST, is it better to use trust or not? 2. If using FRF.12 and told do use compression, does it matter if it is on the WAN DLCI or the policy map? Always in the policy map? 2A If told to use class-based compression then do it on the policy map. 3. SA-SB is 768KB, SA-SC is 1536KB. When applying the 95% of CIR rule do we need to also apply to SC when 1536KB? 3A - What bandwidth threshold would we not use the 95% rule, above 768KB? Thanks, Randall ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] mva using h323 gateway not showing caller name
hi guys, Site A is an h323 gateway along with mva support... when calling from remote destination number to internal phones, the calling name is not showing up on the ip phones...only extension is showing up on the phone..in this case 3001 ip phone displays 2001 number only when 3001 is called after authentication ... is there a way that we can support calling name for mva set up??? Any input is much appreciated. thank you Krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] mva using h323 gateway not showing caller name
dan, on the rdp profile, the extension indeed having the calling name and as well as external number mask... when i call from pstn line(which is remote destinatin number) to any cisco internal ip phone it recognizes the pstn line and shows the internal extension and as well as calling name... but the issue is when 1.) called mva number 2.) authenticated using pin 3.) placed call to 3001 or 4001 etc.. 4.) the call routed succuesfully but shows IP phone displays as 2001 rather showing as 2001 along with calling name SA phone1. 5.) the rdp and extensions of 2001 is already configured with internal ASCII display scenario 2: called from 2024678124 pstn line which is configured as remote destination number, and the call is placed to 77964001 or 4083783001: the display on the ip phone shows with the calling name as well as the extension of 2001. But, the issue is when mva is authenticated the calling name is not supported either the call going out from the pstn gw or to internal cisco phones.. thank you krishna. From: Dan Quinlan (daquinla) daqui...@cisco.com To: Krishna vinayak_...@yahoo.com Cc: Online Study ccie_voice@onlinestudylist.com Sent: Sunday, September 16, 2012 1:47 PM Subject: Re: mva using h323 gateway not showing caller name Go into the DN instance on the remote destination and add the name. It doesn't copy over from DN instance on the hard phone. DQ d...@cisco.com Sent from my iPhone On Sep 16, 2012, at 2:32 PM, Krishna vinayak_...@yahoo.com wrote: hi guys, Site A is an h323 gateway along with mva support... when calling from remote destination number to internal phones, the calling name is not showing up on the ip phones...only extension is showing up on the phone..in this case 3001 ip phone displays 2001 number only when 3001 is called after authentication ... is there a way that we can support calling name for mva set up??? Any input is much appreciated. thank you Krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] mva using h323 gateway not showing caller name
never mind guys.. after skimming the online blog, i see this is expected behavior... the calling name does't show when the phone is ringing, it only shows when the phone is connected.. thank you Krishna. From: Dan Quinlan (daquinla) daqui...@cisco.com To: Krishna vinayak_...@yahoo.com Cc: Online Study ccie_voice@onlinestudylist.com Sent: Sunday, September 16, 2012 1:47 PM Subject: Re: mva using h323 gateway not showing caller name Go into the DN instance on the remote destination and add the name. It doesn't copy over from DN instance on the hard phone. DQ d...@cisco.com Sent from my iPhone On Sep 16, 2012, at 2:32 PM, Krishna vinayak_...@yahoo.com wrote: hi guys, Site A is an h323 gateway along with mva support... when calling from remote destination number to internal phones, the calling name is not showing up on the ip phones...only extension is showing up on the phone..in this case 3001 ip phone displays 2001 number only when 3001 is called after authentication ... is there a way that we can support calling name for mva set up??? Any input is much appreciated. thank you Krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] VoiceView/Service URL
All, please correct that authenticate url is wrong .. the right one is url authenticate http:// CMEipaddress/CCMCIP/authenticate.jsp, not the url authentication http://IP OF CUE/voiceview/authentication/authenticate.do. Thank you krishna. From: Steven Sarrick (ssarrick) ssarr...@cisco.com To: Gurpreet Singh Kukreja tycoononway1...@gmail.com Cc: Online Study (ccie_voice@onlinestudylist.com) ccie_voice@onlinestudylist.com Sent: Friday, September 7, 2012 6:35 AM Subject: Re: [OSL | CCIE_Voice] VoiceView/Service URL Yes, I have these Steven Sarrick SYSTEMS ENGINEER.SALES ssarr...@cisco.com Phone: +1 412 237 6338 Mobile: +1 412 480 3861 Think before you print.This email may contain confidential and privileged material for the sole use of the intended recipient. Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender by reply email and delete all copies of this message. For corporate legal information go to: http://www.cisco.com/web/about/doing_business/legal/cri/index.html From: Gurpreet Singh Kukreja tycoononway1...@gmail.com Date: Friday, September 7, 2012 1:56 AM To: Steven Sarrick ssarr...@cisco.com Cc: Online Study (ccie_voice@onlinestudylist.com) ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] VoiceView/Service URL Hi Steven, - Under Telephony service, i would put the following commands: config# telephony-service url services http://IP OF CUE/voiceview/common/login.do url authentication http://IP OF CUE/voiceview/authentication/authenticate.do The reason for the authentication command above is explained in the document below: http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/rel2_3/cliadmin/ch_vview.html#wp1070594 Configuring Cisco Unified CallManager Express for VoiceView Express: The Authentication Manager is a network server that handles authentication requests for IP phone tasks. The IP phone learns the authentication server URL during the phone's registration process. Cisco Unified CallManager Express (Cisco Unified CME) does not have an authentication server. Cisco Unity Express starts an authentication server that acts as the primary authentication server for VoiceView Express. The Cisco Unified CME administrator must ensure that Cisco Unified CME authentication server URL points to Cisco Unity Express authentication server. The URL format is http://Cisco-Unity-Express-hostname/voiceview/authentication/authenticate.do+ - On Cue, i would make sure of this: service voiceview enable session idletimeout 10 end voiceview - Make sure to do this under Telephony Service: telephony-service no create cnf-files create cnf files Then, reset the phones. Please share your config and we can see what's going on. Regards Gurpreet On Fri, Sep 7, 2012 at 12:03 AM, Steven Sarrick (ssarrick) ssarr...@cisco.com wrote: Practicing VoiceView. Url's are in Telephony Service however the URL's are not being pushed to my phones (go into settings and services/auth url not on phone). When I press services button I get No services configured. End of my lab time so not able to fool with it long. I did reboot, reset phones, basics. Anyone run into this and what is the workaround? Per the DSG for this lab, I did everything as expected. Steven Sarrick SYSTEMS ENGINEER.SALES ssarr...@cisco.com Phone: +1 412 237 6338 Mobile: +1 412 480 3861 Think before you print.This email may contain confidential and privileged material for the sole use of the intended recipient. Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender by reply email and delete all copies of this message. For corporate legal information go to: http://www.cisco.com/web/about/doing_business/legal/cri/index.html ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] VoiceView/Service URL
Kevin, we all under wrong assumption of using url authentication, if anyone used lab 5 workbook, it is clearly cited that url authentication were wrong except for voiceview, in fact the url for authentication should be the CME ip not the CUE ip... when we do integration with via web gui then the correct url were pushed into the telephony-service otherwise those who do cli, they have to remember this command for reference... thank you krishna. From: Kevin Spicer ke...@kevinspicer.co.uk To: Krishna vinayak_...@yahoo.com Cc: Online Study (ccie_voice@onlinestudylist.com) ccie_voice@onlinestudylist.com; Gurpreet Singh Kukreja tycoononway1...@gmail.com; Steven Sarrick (ssarrick) ssarr...@cisco.com Sent: Friday, September 7, 2012 8:58 AM Subject: Re: [OSL | CCIE_Voice] VoiceView/Service URL Krishna, When using voiceview with CME the voiceview authentication url is used in place of the normal CME url. This allows cue to control the phone, needed to play messages in voiceview. On 7 Sep 2012 13:37, Krishna vinayak_...@yahoo.com wrote: All, please correct that authenticate url is wrong .. the right one is url authenticate http:// CMEipaddress/CCMCIP/authenticate.jsp, not the url authentication http://%3C%3Cip/ OF CUE/voiceview/authentication/authenticate.do. Thank you krishna. From: Steven Sarrick (ssarrick) ssarr...@cisco.com To: Gurpreet Singh Kukreja tycoononway1...@gmail.com Cc: Online Study (ccie_voice@onlinestudylist.com) ccie_voice@onlinestudylist.com Sent: Friday, September 7, 2012 6:35 AM Subject: Re: [OSL | CCIE_Voice] VoiceView/Service URL Yes, I have these Steven Sarrick SYSTEMS ENGINEER.SALES ssarr...@cisco.com Phone: +1 412 237 6338 Mobile: +1 412 480 3861 Think before you print.This email may contain confidential and privileged material for the sole use of the intended recipient. Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender by reply email and delete all copies of this message. For corporate legal information go to: http://www.cisco.com/web/about/doing_business/legal/cri/index.html From: Gurpreet Singh Kukreja tycoononway1...@gmail.com Date: Friday, September 7, 2012 1:56 AM To: Steven Sarrick ssarr...@cisco.com Cc: Online Study (ccie_voice@onlinestudylist.com) ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] VoiceView/Service URL Hi Steven, - Under Telephony service, i would put the following commands: config# telephony-service url services http://IP OF CUE/voiceview/common/login.do url authentication http://IP OF CUE/voiceview/authentication/authenticate.do The reason for the authentication command above is explained in the document below: http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/rel2_3/cliadmin/ch_vview.html#wp1070594 Configuring Cisco Unified CallManager Express for VoiceView Express: The Authentication Manager is a network server that handles authentication requests for IP phone tasks. The IP phone learns the authentication server URL during the phone's registration process. Cisco Unified CallManager Express (Cisco Unified CME) does not have an authentication server. Cisco Unity Express starts an authentication server that acts as the primary authentication server for VoiceView Express. The Cisco Unified CME administrator must ensure that Cisco Unified CME authentication server URL points to Cisco Unity Express authentication server. The URL format is http://Cisco-Unity-Express-hostname/voiceview/authentication/authenticate.do+ - On Cue, i would make sure of this: service voiceview enable session idletimeout 10 end voiceview - Make sure to do this under Telephony Service: telephony-service no create cnf-files create cnf files Then, reset the phones. Please share your config and we can see what's going on. Regards Gurpreet On Fri, Sep 7, 2012 at 12:03 AM, Steven Sarrick (ssarrick) ssarr...@cisco.com wrote: Practicing VoiceView. Url's are in Telephony Service however the URL's are not being pushed to my phones (go into settings and services/auth url not on phone). When I press services button I get No services configured. End of my lab time so not able to fool with it long. I did reboot, reset phones, basics. Anyone run into this and what is the workaround? Per the DSG for this lab, I did everything as expected. Steven Sarrick SYSTEMS ENGINEER.SALES ssarr...@cisco.com Phone: +1 412 237 6338 Mobile: +1 412 480 3861 Think before you print.This email may contain confidential and privileged material for the sole use of the intended recipient. Any review
Re: [OSL | CCIE_Voice] Voicemail SRST
steven, for srst voicemail support, just put under the hunt pilot when it explicitly states that alternate extension cannot be used.. thank you krishna. From: Steven Sarrick (ssarrick) ssarr...@cisco.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Monday, September 3, 2012 2:23 PM Subject: [OSL | CCIE_Voice] Voicemail SRST Quick scenario where I find mentions on this list but no firm answer – probably in an archive as I know its out there. Can you point me to a solution or maybe an example in the IPExpert Workbooks of the solution for hitting a voicemail box on UC while in SRST without alternate extension. I'm pretty sure Vik goes over it, but have not seen it in my searches. Steven Sarrick SYSTEMS ENGINEER.SALES ssarr...@cisco.com Phone: +1 412 237 6338 Mobile: +1 412 480 3861 Think before you print.This email may contain confidential and privileged material for the sole use of the intended recipient. Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender by reply email and delete all copies of this message. For corporate legal information go to: http://www.cisco.com/web/about/doing_business/legal/cri/index.html ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] LAN Qos Question
jason, reg: If you leave priority que out configured along with shape of 25 percent then essentially que 1 can have 100 percent of the bandwidth if needs it. Isn't it queue 1 takes whatever the value defined in the threshold i.e. for example mls qos queue-set output 1 threshold 1 138 138 92 138, this takes queue 1 to 138%, and once it is over it comes back to share mode, since the priority queue overwritten the shape, and left over is share value whatever defined at the interface level... please correct me if i m wrong.. thank you krishna. From: murr...@usa.com murr...@usa.com To: murr...@usa.com; Randall Crumm rrcr...@yahoo.com; Dan Quinlan (daquinla) daqui...@cisco.com Cc: ccie_voice@onlinestudylist.com Sent: Saturday, September 1, 2012 8:20 PM Subject: Re: [OSL | CCIE_Voice] LAN Qos Question Priority goes in this order Priority queue out Shape Share If lets say for example you want cos 5 to be in the priority queue but it also states that cos 5 should have no more than 25 percent of the bandwidth. Of course in the mappings you need to put cos 5 in que 1. Then you need to disable priority que out and use shape to give cos 5 only 25 percent (shape 4 0 0 0). If you leave priority que out configured along with shape of 25 percent then essentially que 1 can have 100 percent of the bandwidth if needs it. Make sure you read the question carefully to see what it is wanting. Jason On 9/1/12 at 10:56 AM, Randall Crumm wrote: Hi, I have a question about LAN QoS. If I do shape and share on an interface do I have to disable priority queue out on the interface? Cheers, Randall ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] cisco unity connection doesn't prompt the Sender's ANI
hi guys... when i listen to voicemails for site A or site b phones, it says you have one voicemail from cisco unity messaging system'... i expected this to be Sender's ANI, and for the user's mail box the play back setting are correct that includes the check mark for sender's ANI... but still tells the same prompt... can you advice me where the crucial step i m missing here...any help on this matter is much appreciated. thank you krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] can cupc rtp ports be different with cucm 7 and cucm 8 versions??
hi folks... in the release notes for cupc for cucm 7 version.. i see that rtp ports used are 16384 to 16424, where as the release notes for cucm 8 cites 16384 to 32766.. and so i m little puzzled which one have to taken into consideration ... i think cucm 7 release is incorrect, if ports 16384 to 16424 is used, then technically it can only supports 20 cupc clients isn't it??? 16424-16384 = 40 ports ( 20 odd ports is for RTCP) any advice on this matter is much appreciated... http://www.cisco.com/en/US/docs/voice_ip_comm/cupc/7_0/english/release/notes/ol15710.html thank you krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] can cupc rtp ports be different with cucm 7 and cucm 8 versions??
http://docwiki.cisco.com/wiki/CUPC_ports_for_7.0_to_8.5 i see the port listed differently for the cupc versions in the ciscodoc thank you krishna. From: William Bell b...@ucguerrilla.com To: Krishna vinayak_...@yahoo.com Cc: Online Study ccie_voice@onlinestudylist.com Sent: Thursday, August 30, 2012 11:28 AM Subject: Re: [OSL | CCIE_Voice] can cupc rtp ports be different with cucm 7 and cucm 8 versions?? Krishna, I believe you are misinterpreting the information. The table where you see 16384 to 16424 listed for INBOUND RTP streams is from the CUPC perspective. So, during call setup, CUPC will inform the call processing agent (UCM) that it is ready to receive RTP stream on port X (which will be a number in the range of 16384 and 16424, inclusive). In terms of QoS, if you are asked to deal with port ranges I would use the superset (16384 32767). -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Aug 30, 2012, at 9:01 AM, Krishna wrote: hi folks... in the release notes for cupc for cucm 7 version.. i see that rtp ports used are 16384 to 16424, where as the release notes for cucm 8 cites 16384 to 32766.. and so i m little puzzled which one have to taken into consideration ... i think cucm 7 release is incorrect, if ports 16384 to 16424 is used, then technically it can only supports 20 cupc clients isn't it??? 16424-16384 = 40 ports ( 20 odd ports is for RTCP) any advice on this matter is much appreciated... http://www.cisco.com/en/US/docs/voice_ip_comm/cupc/7_0/english/release/notes/ol15710.html thank you krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] predot trailing # is required to strip for international numbers for sccp phones??
hi guys... i saw kevin's video about call routing section where he discussed about predot trailing # i.e. for a pattern 9.011!#, he said to put the digit discard as predot trailing #... my question is that terminating character has to stripped off while sending the call to gateway??? is it mandatory to do this??? i know for sip phones it is required since rfc 3261 cites the # symbol is no more recognized as terminating character..in sip terminology the # is represented as '%23'... in short, do i have to do predot trailing# for sccp phones if the dial pattern requires that there shouldn't be no interdigit timeout... please advice me on this matter... thank you krishna..___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] mva showing very strange behavior - never routing the calls to mobile
hi folks... i encountered a very strange behavior with mva configuration... i configured mva for h323 site with number 3000, and in the service parameters i enable mva along with mva number, and as well as partial match and 7 digits... remote destination profile has mobile and mobile connect enabled, and the remote destination number is given as 98397263 incoming call from pstn has 10 digit ani display i.e. 4088397263 ... 1.) first weird behavior : mva didn't recognizes this number, and prompts for enter the remote destination number 2.) call made from 2001 to 3001 only rings 3001 but no traces that it hits h323 gateway in routing the call to mobile number... i enabled both debug voip dialp and deb isdn q931 Now, i changed the remote destination number to 4088397263..then it recognizes this number and asks for pin... with the above configuration, here are the weird results.. when i call to 3001 from pstn phone, it sends the call directly to voicemail rather ringing the extension 3001,and the call is staying for 5 seconds then terminates itself automatically... i am completely lost when this happened.. and when i made the remote destination number to 98397263 it works fine i.e. the 3001 phone rings ... Did anyone experienced this type of behavior in your practice labs.. any advice or help on this matter is much appreciated.. note: i placed mobile access partition in a different partition that only css of h323 gateway sees it, and also placed in null partition but no difference .. thank you krishna. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] mva showing very strange behavior - never routing the calls to mobile
jason, from router to ccm the dnis is 4 digits whereas ANI is sent completely to cucm whatever the pstn is delivering... From: Jason Aarons (AM) jason.aar...@dimensiondata.com To: Krishna vinayak_...@yahoo.com; Online Study ccie_voice@onlinestudylist.com Sent: Wednesday, August 29, 2012 6:36 AM Subject: RE: [OSL | CCIE_Voice] mva showing very strange behavior - never routing the calls to mobile How many digits are you sending from router to ccm? 4? Or 7? Show run | begin dial-peer ? Debug voice translation rule Debug voip dialpeer From:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Krishna Sent: Wednesday, August 29, 2012 2:19 AM To: Online Study Subject: Re: [OSL | CCIE_Voice] mva showing very strange behavior - never routing the calls to mobile hi folks... i encountered a very strange behavior with mva configuration... i configured mva for h323 site with number 3000, and in the service parameters i enable mva along with mva number, and as well as partial match and 7 digits... remote destination profile has mobile and mobile connect enabled, and the remote destination number is given as 98397263 incoming call from pstn has 10 digit ani display i.e. 4088397263 ... 1.) first weird behavior : mva didn't recognizes this number, and prompts for enter the remote destination number 2.) call made from 2001 to 3001 only rings 3001 but no traces that it hits h323 gateway in routing the call to mobile number... i enabled both debug voip dialp and deb isdn q931 Now, i changed the remote destination number to 4088397263..then it recognizes this number and asks for pin... with the above configuration, here are the weird results.. when i call to 3001 from pstn phone, it sends the call directly to voicemail rather ringing the extension 3001,and the call is staying for 5 seconds then terminates itself automatically... i am completely lost when this happened.. and when i made the remote destination number to 98397263 it works fine i.e. the 3001 phone rings ... Did anyone experienced this type of behavior in your practice labs.. any advice or help on this matter is much appreciated.. note: i placed mobile access partition in a different partition that only css of h323 gateway sees it, and also placed in null partition but no difference .. thank you krishna. itevomcid ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] five lab workbook CUCM-DHCP issue
hi folks... So far i have done 6 labs practicing ipexpert five lab workbook, and everytime i encountered the CUCM dhcp issue where site A and Site B phone are unable to register due to dhcp issue... on site A router A, and Site B router interface's had ip helper-address 10.10.210.10.. when i debup ip packet for acl that includes udp ports 67 68.. here is the message i found and tried google it but no luck in finding the solution... here is the debug that looks like.. FIBipv4-packet-proc: packet routing failed Note: the dhcp configured local to router's work fine with no issues... thank you krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] SLRG is same as LRG???
hi folks, i was wondering slrg and lrg are same in terms of functionality... can both viewed as same ..??? please advise me on this query... thank you krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] calling name support voicemail in srst call-man-fallback
hi folks, for a srst site with call-manager-fallback supports calling name, and when i call the voicemail from the srst phone it doesn't play any voicemail instead it plays CUC general message, is this expected behavior with call-manager-fallback pstn callers are able to leave a voicemail to this phone while in srst... any advice on this matter is appreciated.. thank you krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] calling name support voicemail in srst call-man-fallback
bill, the point is pstn callers are able to leave voicemail when the site is in SRST, the real problem is retrieving the message from the CUC... i don't understand what does it has to do with alternate extension.. thank you krishna. From: Bill Lake whl...@gmail.com To: Krishna vinayak_...@yahoo.com Sent: Friday, August 24, 2012 9:14 AM Subject: Re: [OSL | CCIE_Voice] calling name support voicemail in srst call-man-fallback did you set up an alternate extension in CUC? On Fri, Aug 24, 2012 at 8:02 AM, Krishna vinayak_...@yahoo.com wrote: hi folks, for a srst site with call-manager-fallback supports calling name, and when i call the voicemail from the srst phone it doesn't play any voicemail instead it plays CUC general message, is this expected behavior with call-manager-fallback pstn callers are able to leave a voicemail to this phone while in srst... any advice on this matter is appreciated.. thank youkrishna. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CUE jtapi doesn't register via cli
hi folks, as everyone suggesting that one should be familiar with cli command for cue setup and apparently i am following the same path suggested by experts... i did ccn trigger, ccn application etc and when i look into cucm the cti rp and cti ports were never registered, and so i went through gui where i was asked to put the web administrator userid and password which is an additional task compared to cli configuration, and after everything is done and saved the config with reload at final step...viola cti port and cti rp both shows as registered... This really took double time in term of reconfiguration and web gui access... can anyone advice what i m missing in my cli approach... note: after doing the cue configuration using cli, i saved the config and reloaded the cue before approaching the gui process... any advice is much appreciated. thank you krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] cme as srst voicemail showing internal extension rather pstn number for active xfer call
hi folks, for site c when it is in srst mode, i made the call to sc phone1 and transferred the active call via transfer button and left a voice mail.. when i listen to voicemail it prompts the internal extension as the caller rather prompting as an unknown caller.. when it is CUCM mode, the voicemail prompts shows up correctly but in srst its not... is there any command that needs to be enabled in the cue for active transfer calls?? thank you krishna..___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] mwi ephone-dn required for mgcp with cme as srst???
hi folks, i'd like to know whether we need to have create ephone-dn for mwi on and off for mgcp gateway in cme srst mode... the cue is integrated with cucm, but in fall back mode i was wondering if we have to create mwi dial-peers ??? In srst mode, i configured dial-p voip , and sip-ua and mwi server, and on ephone-dn mwi sipapart from this configuration do i required to create ephone-dn for mwi as well/? thank you Krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] subscribe cucm is not showing up
hi james/doug i don't see cucm subcriber showing up on the switch , and also i cannot reach the ip address of the subscriber as well... CUC7-Pub Fas 1/0/4 144 H none foun eth0 SiteA-RTR Fas 1/0/1 179 R S I 2811 Fas 0/0 UCMPub Fas 1/0/4 130 H none foun eth0 SEP001BD4C6C195 Fas 1/0/2 178 H IP Phone Port 1 UCCX7-PUB. Fas 1/0/4 177 H Win2000 S Eth 2/2 can you please fix this ??? thank you krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] subscribe cucm is not showing up
any update would be much appreciated.. From: Krishna vinayak_...@yahoo.com To: Online Study ccie_voice@onlinestudylist.com Sent: Tuesday, August 14, 2012 5:32 PM Subject: subscribe cucm is not showing up hi james/doug i don't see cucm subcriber showing up on the switch , and also i cannot reach the ip address of the subscriber as well... CUC7-Pub Fas 1/0/4 144 H none foun eth0 SiteA-RTR Fas 1/0/1 179 R S I 2811 Fas 0/0 UCMPub Fas 1/0/4 130 H none foun eth0 SEP001BD4C6C195 Fas 1/0/2 178 H IP Phone Port 1 UCCX7-PUB. Fas 1/0/4 177 H Win2000 S Eth 2/2 can you please fix this ??? thank you krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] wan Qos :police or bandwidth
hi folks, when doing wan qos, i come across the situation where signalling has to get 32kbps and in the question it didn't said anything about policing...and therefore my question is which command should we have to use to accomplish this task... i used bandwidth 32 under class-map which is called from the policy-map... and also another way of doing it which is police 32000 under class-map is it really matters when using police or bandwidth for the wan qos...??? plz advice me on this matter... thank you krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] ntp master- is it necessary
This email and any files transmitted with it are confidential and are intended for the sole use of the individual to whom they are addressed. Black Box Corporation reserves the right to scan all e-mail traffic for restricted content and to monitor all e-mail in general. If you are not the intended recipient or you have received this email in error, any use, dissemination or forwarding of this email is strictly prohibited. If you have received this email in error, please notify the sender by replying to this email. -- Message: 2 Date: Sun, 5 Aug 2012 10:14:44 -0300 From: Bruno Nonogaki brun...@gmail.com To: Krishna vinayak_...@yahoo.com Cc: Online Study ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] ntp master- is it necessary Message-ID: CAP_RLdVVgHfN=fGytf6w7K5n5-nS6g3JnRiZfM5=tgsxozp...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hello Krishna, Yes, you are right. ntp master is not required. If you do ntp master, it may synchronize with its internal clock. It is a big mistake a lot of people do, including me before the OWLE Bootcamp, which I really recommend. Regards, Bruno On Sun, Aug 5, 2012 at 2:37 AM, Krishna vinayak_...@yahoo.com wrote: hi folks, i see some guys posts on ntp master command on the hq router ... i was wondering why one would be needing ntp master command when it is already being synchronized with external ntp server ntp master will infact mess up the time if not configured correctly since ntp master takes the stratum from the hardware(device) and be careful when putting the command ntp master .. if it is required then it is advised to keep the stratum number high compared to the extrenal ntp server... please correct me guys if i m wrong precisely, i felt that ntp master command is not required if that device is synchronized with external ntp server.. any comments on my advice is much appreciated... thank you krishna. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20120805/f5a0f2f7/attachment-0001.ht ml -- Message: 3 Date: Sun, 5 Aug 2012 10:21:30 -0300 From: Bruno Nonogaki brun...@gmail.com To: Justin McIntyre justin.mcint...@blackbox.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Switch QOS query Message-ID: cap_rldxltmb2zgsretwit5xz5aiq7bxgwvooygfg1xvwugh...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi Vir, I agree with Justin regarding the issue with the requirements. And I also recommend you the Kevin Wallace's video: http://www.youtube.com/watch?v=IA4iOrn2eiU Regards, Bruno On Sun, Aug 5, 2012 at 10:10 AM, Justin McIntyre justin.mcint...@blackbox.com wrote: So I believe your on the right track with your QOS config but there are a few things that need to be modified. 1. I see an issue with your requirements. Have the priority-queue enabled but then also give queue 1 30% bandwidth. If priority-queue out is enabled then this over-rides the bandwidth command for that queue. I know you had some other questions as well specifically about how to drop certain traffic if a queue were 80% full. My suggestion to you would be to review Vik Mahlis QOS blog on the IPEXPERT website. Go to blog.ipexpert.com and select the voice blog on the left. Then look for the QOS section. I think this will clear up most of your questions and get you on your way. Thanks, Justin McIntyre This email and any files transmitted with it are confidential and are intended for the sole use of the individual to whom they are addressed. Black Box Corporation reserves the right to scan all e-mail traffic for restricted content and to monitor all e-mail in general. If you are not the intended recipient or you have received this email in error, any use, dissemination or forwarding of this email is strictly prohibited. If you have received this email in error, please notify the sender by replying to this email. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20120805/9a629423/attachment-0001.ht ml
Re: [OSL | CCIE_Voice] is it mandatory to have h323 dial-peer for bacd
hi folks, i have site c which is an mgcp gateway, and during srst it has to support bacd for call queuing mechanism... and therefore i invoked the hunt pilot on the incoming pots dial-peer... the call came through the pots dial peer, and the welcome greeting is played back after which the hunt pilot number extensions started ringing... my query is do i need to have h323 voip dial peer as well for the bacd setup... i am not sure about this and therefore would like to get some feedback from you guys... any help on this matter is much appreciated.. thank you krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] no contacts showing up in CUPS
Hi folks, this query is about CUPC clients CUP server... I enable one of the site phone as softphone and the other one as desk phone mode... everything works fine including voicemails... but the problem is in order to send the message to other client (softphone/deskphone) i need to add them in the contact list but when i try to add them i don't see any contact list popping up in the search ... i am attaching the screen shot for reference with this email.. | can anyone advice me what could be the cause for not showing the contacts ??? your help is appreciated.. thank you krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] no prompts with bacd
hi folks, i ran into a new issue with the bacd.. but with your help i could able to get the welcome prompt and i understand the bacd structure... here is my bacd script, where i called the 3000 number, and it gives the welcome prompt followed by extensions 3001 , 3002 ringing with gap interval of 10 seconds... both extensions didnt answer the call and it played the busy prompt to the pstn user, but here is the problem after the busy prompt the call is getting cleared/hung rather waiting in the call queue and update the queue number 2 or 3 etc please correct my script if i did miss any crucial steps that allows the pstn call to be on hold, and update the queue accordingly...for instance when both extensions didn't answer the call it shows as queue 1.. and after that busy prompt played back to pstn and i was expecting the call queue will increase to 2 but the call got hung up... application service app-b-acd param queue-manager-debugs 1 param aa-hunt1 3000 param number-of-hunt-grps 1 param queue-len 10 service app-b-acd-aa paramspace english location flash: paramspace english index 0 paramspace english language en param aa-pilot 3000 param number-of-hunt-grps 1 param service-name app-b-acd param handoff-string app-b-acd-aa param second-greeting-time 20 param drop-through-option 1 param drop-through-prompt _bacd_welcome.au param call-retry-timer 60 param max-time-call-retry 700 param voice-mail 3220 param max-time-vm-retry 2 any advice on this matter is much appreciated.. thank you krishna. From: Peter Simmons pe...@grayrigg.com To: Krishna vinayak_...@yahoo.com Sent: Saturday, August 11, 2012 2:51 AM Subject: Re: [OSL | CCIE_Voice] no prompts with bacd Krishna, You config has a couple of errors around the prompt parameters. 1) Your configured parameter drop-through-prompt (with value of _dt_prompt.au) matches a file on flash called en_dt_prompt.au (based on the parameter you configured english language en). This prompt file is not supplied as part of the B-ACD system files - if you need this, record your own, or substitute/copy the welcome prompt file, give it the name you have set this parameter to, and it will be called at the beginning of the call flow. There are rules around filenames versus parameter values because of the language prefix - (en in this case) - that may trip you up if you don't spot the trap. If you hear no greeting at the beginning of a drop-through call, then this parameter is most likely incorrectly set, or the filename you have set this to does not exist. For your scenario, I would set this parameter to the welcome prompt file en_bacd_welcome.au since this has the correct wording, and should exist on flash as part of the default setup: param drop-through-prompt _bacd_welcome.au 2) There is no parameter called welcome in the default script that refers to a welcome prompt, you need param welcome-prompt _bacd_welcome.au if you want to set this - but parameter isn't used anyway for drop-through calls, it plays the prompt associated with drop-through-prompt instead . You can probably delete this line. 3) The parameter you created busy is not going to be referenced by the default script, so it won't do anything. You can probably delete this line. 4) If you have the file en_bacd_allagentsbusy.au on flash, then this will be played periodically whilst calls are in the queue - this is the second greeting associated with the second-greeting-time parameter. You cannot change the name of this file in the default built-in script. The documentation isn't brilliant, but it's got some helpful examples and notes about prompts etc. http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html Vik has written a great blog on this, worth a read! http://blog.ipexpert.com/2009/01/24/b-acd-in-a-nutshell/ regards Peter On 11/08/2012 05:22, Krishna wrote: hi folks, i configured on site C cme with the bacd script using the cisco configuration example guide... but unfortunately it didnt work for me while i try to establish the requirements.. the requirement is 1.) thank you or welcome prompt 2.) call route to ephone-hunt group 4000 3.) if both phones are busy then it should play busy prompt here is my configuration: application service app-b-acd-aa param voice-mail 4110 paramspace english index 0 param max-time-call-retry 700 param service-name app-b-acd param number-of-hunt-grps 1 param drop-through-option 1 paramspace english language en param handoff-string app-b-acd-aa param max-time-vm-retry 2 paramspace english location flash: param aa-pilot 4000 param second-greeting-time 60 param drop-through-prompt _dt_prompt.au param busy _bacd_allagentsbusy.au param welcome _bacd_welcome.au param
[OSL | CCIE_Voice] no prompts with bacd
hi folks, i configured on site C cme with the bacd script using the cisco configuration example guide... but unfortunately it didnt work for me while i try to establish the requirements.. the requirement is 1.) thank you or welcome prompt 2.) call route to ephone-hunt group 4000 3.) if both phones are busy then it should play busy prompt here is my configuration: application service app-b-acd-aa param voice-mail 4110 paramspace english index 0 param max-time-call-retry 700 param service-name app-b-acd param number-of-hunt-grps 1 param drop-through-option 1 paramspace english language en param handoff-string app-b-acd-aa param max-time-vm-retry 2 paramspace english location flash: param aa-pilot 4000 param second-greeting-time 60 param drop-through-prompt _dt_prompt.au param busy _bacd_allagentsbusy.au param welcome _bacd_welcome.au param call-retry-timer 15 ! service app-b-acd param queue-len 10 param aa-hunt1 4000 param queue-manager-debugs 1 param number-of-hunt-grps 1 can anyone tell me what _dt_prompt.au stands for... i dont this .au file anywhere on the cme router flash... and moreover param weclome and param busy are the one created by me in the application but no use since it didnt work.. any help on this matter is much appreciated... thank you krishna. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] MVA -Your call can not be completed as dialed
Rat, make sure mva is enabled on the service parameters and number as well in the cucm service parameters, and also check with the dial-peer and application url on the router with the right number... Vipul. it uses rerouting css when it makes outbound calls, but in this case he can't even get to the prompt of mva... thank you krishna. From: Vipul Jindal (vipjinda) vipji...@cisco.com To: ccielabrat ccielab...@gmail.com; ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Tuesday, August 7, 2012 2:29 PM Subject: Re: [OSL | CCIE_Voice] MVA -Your call can not be completed as dialed It uses the re routing CSS on the remote destination number. If you check the call manager traces, you can easily check it. From: ccielabrat ccielab...@gmail.com Date: Tuesday, August 7, 2012 2:09 PM To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] MVA -Your call can not be completed as dialed To All, I'm hoping the group can help me understand the call flow for an MVA call. I'm able to call into the MVA pilot number , have my remote destination number recognized and be prompted for my PIN and to dial . But I get the message Your call can not be completed as dialed for anything I try to call. I understand that the number configured under the mobile voice access page is used as an anchor , as per Vik's vlecture, but I'm unclear what device is referenced regarding CSS and what should and should be reachable. Can anyone please help get closure on this last piece of the puzzle. -Lab Rat ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Qos lan egress queue no phones registering via cucm dhcp
hi folks, this query is about lan qos and dhcp config on cucm, yesterday i had an issue where i couldn't able to register the Site A and Site B phones to cucm. Cucm is the dhcp server, and initially i put the default route ip address in the primary dns field in the dchp subnet, later corrected it by placing it in the primary router ip address, and after dhcp monitor service is restarted... but phones are not registering to the cucm... i did debug ip dhcp server event/packet, and i do see the dhcp relay is passing the info to helper address on the Site B. But, it says that no dhcp pool for the site B, whereas it does have the dhcp pool in the cucm... this is all about Site B.. Site A: one of the phone registered to CUCM but the ip address is out of the dhcp scope... i shut down the port on the switch, and no shut but no luck Qos query: i am little confused about the egress queue on the lan... is the egress queue applies only to the uplink port i.e. port connecting to router??? or is that we have to view each single port has 2 ingress queue and 4 egress queue as combined??? can anyone help me on this matter.. your help is much appreciated.. any advice or help is much much appreciated, since i spent 7 hours troubleshooting the phone registration but couldn't crack it... thank you krishna..___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] ntp master- is it necessary
hi folks, i see some guys posts on ntp master command on the hq router ... i was wondering why one would be needing ntp master command when it is already being synchronized with external ntp server ntp master will infact mess up the time if not configured correctly since ntp master takes the stratum from the hardware(device) and be careful when putting the command ntp master .. if it is required then it is advised to keep the stratum number high compared to the extrenal ntp server... please correct me guys if i m wrong precisely, i felt that ntp master command is not required if that device is synchronized with external ntp server.. any comments on my advice is much appreciated... thank you krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] srst - voicemail for cue intergrated with cucm
hi folks, i have a site C with cue integrated with cucm, and the site C phones are registered to call manager as well. when site operates in SRST, how can i able to make voicemail to work, since CUE is integrated with cucm... Does CUE supports both ccm and cue features or the voicemail doesn't work in srst mode?? any advice or suggestion on this query is much appreciated.. thank you krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] no voice recording using phone via UC
hi folks, I think this is my 6 or 7 attempt in Unity connection for recording the voice prompts using UC greetings... i can able to record and retrieve using computer, whereas using the phone it doesn't do anything, and the fields are graded out... any thoughts how to make this work using proctorlabs... your help is much appreciated... thank you krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] service url in the cucm for the phoneview
hi folks, can any one help me out with the service url for phoneview for ip phones in cucm thank you krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] SRST Access-list for home equipment
Rrcrumm, apply this acl on the inbound interface when you do this step, then there is no need adding additonal acl statements mentioned by Dan... remember you applied it on the outside interface which doesn't have any control in regulating the remote host with exception adding the Dan acl also to make it work... Dan, when ccm-manager fall-back mgcp command is used, and under telephony-service the command srst ephone description doesn't show up on the phone..rather it showed as Cisco Cme, whereas the description is given as your current options... when i gave the command system message then it showed as your current options.. is this is a bug in srst command under telephony service??? thank you krishna. From: Rrcrumm rrcr...@yahoo.com To: Dan Quinlan (daquinla) daqui...@cisco.com Cc: Online Study ccie_voice@onlinestudylist.com Sent: Sunday, July 29, 2012 9:58 PM Subject: Re: [OSL | CCIE_Voice] SRST Access-list for home equipment Thanks Dan I'll try that Sent from my iPhone On Jul 29, 2012, at 7:52 PM, Dan Quinlan (daquinla) daqui...@cisco.com wrote: Oh and I'd apply the access group on interface vlan 12 (the phone vlan) in both directions ip access group sc in and up access group sc out DQ d...@cisco.com Sent from my iPhone On Jul 29, 2012, at 10:48 PM, Dan Quinlan (daquinla) daqui...@cisco.com wrote: You need to add rules for the other direction as well (pub and sub to the phone). Otherwise the phone still receives keepalives. So you need to add these to your access list: deny ip host 10.10.210.10 host 192.168.12.12 deny ip host 10.10.210.11 host 192.168.12.12 DQ d...@cisco.com Sent from my iPhone On Jul 29, 2012, at 10:40 PM, Randall Crumm rrcr...@yahoo.com wrote: Hello, I am working on PL but with my equipment. I want to make the phones here go into SRST. SO I need to add an access-list, my hoe phone being IP address 192.168.12.12 So I added this ip access-list extended sc deny ip host 192.168.12.12 host 10.10.210.11 deny ip host 192.168.12.12 host 10.10.210.10 permit ip any any Then applied it to the interface: interface FastEthernet0/0 description (Outside Public Interface) ip address dhcp ip access-group sc out no ip unreachables ip mtu 1400 ip nat outside ip virtual-reassembly duplex auto speed auto no cdp enable crypto ipsec client ezvpn Voice-vRack This is not working. Any thoughts? Cheers, Randall ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] ANI/DSNI TON on Proctor Labs
if it is h323 gateway, i will create a translation rule and apply at the dial-peer... if it is mgcp gateway do it on the call manager route list detail level... thank you krishna. From: Randall Crumm rrcr...@yahoo.com To: Online Study ccie_voice@onlinestudylist.com Sent: Monday, July 30, 2012 11:33 AM Subject: [OSL | CCIE_Voice] ANI/DSNI TON on Proctor Labs Hello, I have noticed some different behaviors and was wondering what you recommend If the question asks for plan unknown and type unknown should you set to 1. plan unknown and type unknown or 2. plan isdn type unknown or 3. plan call manager type call manager For me I have tried the above and it seems like call manager/call manager is what is working(actually allowing the call to go through). It goes through as unknown/unknown Any thoughts? Thanks! Randall ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] no ipcc extension showing up
hi folks, i configured ippa service url under the phone services.. and i subscribed to the phone as well.. but when i browsed the end user to subscribe the ipcc extension, i dont see ipcc extension at all.. i even restarted the cm services but no luck... any advice on this matter is much appreciated. thank you krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] no ipcc extension showing up
Thanks bill dan helped me out on this perspective.. thanks once again bill for your quick response.. From: Bill Lake whl...@gmail.com To: Krishna vinayak_...@yahoo.com; ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Saturday, July 28, 2012 9:58 PM Subject: Re: [OSL | CCIE_Voice] no ipcc extension showing up You can find this at the following path www.cisco.com/support Scroll to bottom of page Click configure and this takes you to the page if you are taking the lab http://www.cisco.com/cisco/web/psa/default.html Now click on products - Voice and Unified Communications - Customer Collaboration - Cisco Unified Contact Center Products - Cisco Unified Contact Center Express Select configuration examples and technotes Search For Callmanager Click on ICD Extension Option Does Not Appear on the Cisco CallManager Global Directory User Page You will find the sql statement at the bottom, just adjust it for your needs. run sql update processconfig set paramvalue=T where paramname like '%nstalled%' From: Krishna vinayak_...@yahoo.com Reply-To: Krishna vinayak_...@yahoo.com Date: Saturday, July 28, 2012 9:26 PM To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] no ipcc extension showing up hi folks, i configured ippa service url under the phone services.. and i subscribed to the phone as well.. but when i browsed the end user to subscribe the ipcc extension, i dont see ipcc extension at all.. i even restarted the cm services but no luck... any advice on this matter is much appreciated. thank you krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] IPPM Audio and Visual Alert
check service url authentication in the service parameters to the pub ip address... From: Juan Carlos Anzola juancarlosanz...@gmail.com To: ccie_voice@onlinestudylist.com Sent: Thursday, July 26, 2012 4:26 PM Subject: [OSL | CCIE_Voice] IPPM Audio and Visual Alert Hi Guys, I am configuring IPPM, everything is working fine, except i am not getting the visual and audio alert when recieving a message. I have checked the following: * IPPM user and password match between CUCM and CUPS * IP Phone configured with Owner User ID * End User associated with IP Phone * Line Appearence associated with End User What i am missing here? Regards, -- Juan Carlos Anzola ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] no voicemails in the mailbox when leaving voicemails from pstn line br2 sire
hi folks, i've experienced a very strange behavior with Unity connection, where i was redirected to voicemail of Hq site phones when called from the pstn or br2 site, and i could able to leave a voice mail to the mailbox of the the user, but strangely no mwi lit up in addition to no voicemail is showing up on the user's mail box. Inititally i thought it might be a transcoding issue, but i verified that when i recorded the message on pstn phone i pressed # symbol and i could able to listen to my recorded message for verification. But after leaving the voicemail, i cannot see or hear that voicemail in the mail box of Hq site phone 1. When i call from br1 site phone to hq site phone, the mwi works and also voicemails are also showing up with no issue. the real issue is when calling from branch 2 or pstn phone the voicemails are not showing up on the user's mailbox along with mwi functionality, though it is redirected to voicemail .. All these sites are registered to the call managers, where Hq, Br2 are the mgcp gateways whereas br1 is h323 gateway.. any advice or help on this issue is much appreciated thank you krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] no notification sound on ip phone for messages
hi folks, i have enabled IPPM service on the cucm, and subscribed them to devices. I can able to login into IPPM service phone, and can able to view the messages after clicking on the contact, but i don't see the updated messages until i exit and then click on the messages option again on ipphone, and moreover there is no sound notification for the chat message from other phones. Is this is expected behavior??? I can hear the sound notifications on cupc whereas on ip phones are not... any advice on this matter is much appreciated. thank you krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] [OSL|CCIE Voice] no notification sound on ip phone for messages
Bruno Justin, I did enabled all these and even verified twice... what authentication url has to be there on enterprise parameters??/ thank you krishna. From: Bruno Nonogaki brun...@gmail.com To: Justin McIntyre justin.mcint...@blackbox.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Monday, July 23, 2012 4:39 PM Subject: Re: [OSL | CCIE_Voice] [OSL|CCIE Voice] no notification sound on ip phone for messages And also check the Authenticate URL on Enterprise Parameters... On Mon, Jul 23, 2012 at 5:44 PM, Justin McIntyre justin.mcint...@blackbox.com wrote: this means you do not have all configuration completed. You need to check these few places: 1. user licensed for CUP in UCM 2. You have created the Application user for IPPM(PhoneMessenger) in UCM and the phone you are using the IPPM service on is associated with this Application user.. Also make sure this user is CTI enabled and that the passwords in UCM and Application IPPM are the same, also make sure the IPPM status is set to on. Additionally if you want to see presence updates make sure you have your SIP trunk from UCM to CUPS set properly and that the user that you want to see presence updates from has been associated with the line/DN. This email and any files transmitted with it are confidential and are intended for the sole use of the individual to whom they are addressed. Black Box Corporation reserves the right to scan all e-mail traffic for restricted content and to monitor all e-mail in general. If you are not the intended recipient or you have received this email in error, any use, dissemination or forwarding of this email is strictly prohibited. If you have received this email in error, please notify the sender by replying to this email. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] setup for mgcp pri monitoring
hi folks, i do remember some member asking for monitor status of mgcp pri when it is in the state of TEI 0 changed to down... here are the detail steps: 1.) open the RTMT Tool, right click on the performace under performance section 2.) under cluster, choose the call manager to which mgcp is registered 3.) extend the call-manager section by a one click on the call-manager IP address 4.) Choose the CISCO MGCP PRI DEVICE 5.) select DataLinkInService tab, then right click on this tab and select the counter instances 6.)on the Right pane of the window, the performance monitor appears 7.)on performance monitor, select and right click which gives the option of set Alert/properties 8.) under Set alert/properties a.) select the severity type b.)define threshold value, in this case check under tab with value 1 since 0 is down , 1 value is UP C.)frequency setup for monitoring d.) check enable email, click on the configure tab, add a new alert action . 9.)under serial interface issue no isdn bind-l3 ccm-mana or shutdown the interface which automatically brings the D channel out of service... Note: TEI Assigned is nothing but D-channel is not active, and therefore it would meet the requirement for the alert action please correct me if i am missing something or my approach is incorrect.. thank you krishna. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] dead air when calling CUE vmail gui doesn't work
hi folks, I couldn't able to understand why the CUE giving me the dead air though after the configuration is absolutely correct with the right codecs. when i pressed the vmail button on the phone, it connects to the vmail number but i cannot hear anything. And, also i couldn't access web gui for the cue even after providing all the right info such as ip http server, ip http path, ip http auth local.. the web browser sits there forever with no output... does anyone experienced the same problem as i am??? your advice on this matter is much appreciated. thank you krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] gateway config is on lo 0 but voice sub interface showing as gateway
hi folks, i couldn't understand the logic behind the configuration for the vol2 lab 1 gatekeeper section invoking cube functionality. here is the call flow: cucm (gateways) --- gatekeeper(hq lo 0) cme (gateway) gatekeeper (hq lo 0) CUBE (gateway) gatekeeper (hq lo 0) when i call from cucm phone to cme phone, the call shows as 2 i.e from cucm to cube, and from cube to cme. I am puzzled with the configuration on the ipexpert soultion guide. on HQ int lo 0 i.e. 10.10.110.1 here is the config: int lo 0 h323-gateway voip h323-gateway voip id UCM 10.10.110.1 1719 h323-gateway voip h323-id CUBE So, technically the lo 0 is having the gateway configuration, but the h323-gateway bind is on the int fa 0/0.20 sub interface of hq-router. isn't the call flow should be CUCM to cube(10.10.110.1), and then cube(10.10.110.1) to cme. but the call flow is appearing as: cucm 10.10.200.3 10.10.200.3 -- cme. how could it be possible to have 10.10.200.3 acting as gateway having the gateway config on interface lo 0. any advice on this matter is much appreciated. thank you krishna. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] mlp vs frf.12 when to use ???
hi folks, from my observations, i found that mlp would be used if and only if, there are two sites connecting to a single site via the same physical interface. And, for frf.12 we use it for ppp link. please correct me or advise me if i m wrong. Thank you krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] transcoder not functioning in the cube section
i did enable the faststart but no use, and also this transcoder is locally available to cube as well... From: Lidiya Krunic lkru...@hotmail.com To: luv...@gmail.com; vinayak_...@yahoo.com Cc: ccie_voice@onlinestudylist.com Sent: Wednesday, June 20, 2012 10:40 PM Subject: RE: [OSL | CCIE_Voice] transcoder not functioning in the cube section Try to remove checkmark Wait for Far End TCS on CUM (or use faststart). Date: Thu, 21 Jun 2012 08:37:54 +0530 From: luv...@gmail.com To: vinayak_...@yahoo.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] transcoder not functioning in the cube section Transcoder should be available locally for cube On Jun 21, 2012 8:23 AM, Krishna vinayak_...@yahoo.com wrote: hi folks, I got stuck at vol2 lab1 in section 4.2, where the cube involves in routing the calls using gakekeeper. when i call from 1002 to 3002 the phone rings, and when i answer the 3002, the 1002 still makes the ringing sound, and after some time the call failing with the busy tone. I checked the show sccp connections, and surprisingly it is not showing any transcode sessions. when i call from 1002 or 5002 to sip phone 3006, it gives me immediately busy-tone/fail tone. can you guys advice me what to do in order to make this work. here is my config: HQ-RTR(config)#do sh sdspf unit mtp-1 Device:hqgk-xcode TCP socket:[1] REGISTERED in SCCP ver 65546/10 actual_stream:6 max_stream 6 IP:10.10.200.3 51291 MTP Dixieland keepalive 152 Supported codec: G711Ulaw G711Alaw G729 G729a G729b G729ab max-mtps:2, max-streams:6, alloc-streams:6, act-streams:0 HQ-RTR(Config)#dial-peer voice 3000 voip incoming called-number 3...$ dial-peer voice 3001 voip destination-pattern 3... session target ras codec g711ulaw gatekeeper zone local UCM ipexpert.com zone local UCME ipexpert.com outvia VIA zone local VIA ipexpert.com zone prefix UCM 1... gw-priority 10 gk-trunk_2 zone prefix UCM 1... gw-priority 9 gk-trunk_1 zone prefix UCME 3... zone prefix UCM 5... gw-priority 10 gk-trunk_2 zone prefix UCM 5... gw-priority 9 gk-trunk_1 gw-type-prefix 1#* default-technology no shutdown HQ-RTR(config)#do sh run | s gatew h323-gateway voip interface h323-gateway voip id VIA ipaddr 10.10.110.1 1719 h323-gateway voip h323-id HQ-RTR h323-gateway voip bind srcaddr 10.10.200.3 gateway Thank you krishna. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 76, Issue 42
Thanks to Justin , Randell for your awesome help. I will try this setting in my next lab, and let you know with the updates. Thanks a lot once again for all of you.. Krishna. From: Justin McIntyre justin.mcint...@blackbox.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Thursday, June 14, 2012 7:55 AM Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 76, Issue 42 Randall is correct here. UCM will always divert to the intra-region Service Parameter settings. Change this to 729 and then hard code your codec between regions within the region parameters section. Thanks, Justin McIntyre Engineer Mutual Telecom Services Inc. a wholly-owned subsidiary of Black Box Corp. COMM: (434)-946-1562 DSN: (312)-237-1562 CELL: (540)-312-9391 FAX: (434)-946-1510 -Original Message- Message: 6 Date: Thu, 14 Jun 2012 00:14:03 -0700 From: Rrcrumm rrcr...@yahoo.com To: Krishna vinayak_...@yahoo.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Gatekeeper acting weird for codec Message-ID: 0d193090-0965-4053-9d15-0a2d160ea...@yahoo.com Content-Type: text/plain; charset=us-ascii I think I remember some tone a k this situation. You chance the setting is service parameters for intra region to g729 HTH Randall Sent from my iPhone On Jun 13, 2012, at 9:10 PM, Krishna vinayak_...@yahoo.com wrote: Hi folks, I configured the gatekeeper on the Hq router, and when i call from hq to br2(cme) the call set up shows as 16 kbps, but whereas from cme to hq it shows as 128 kbps but the actual call is connected with g729. Even after the call got connected, it stills shows as 128 in the show gatekeeper call. here is the output status: i put the gk-trunk in the hq region as well. any help is much appreciated on this matter. BR2-RTR(config)#do sh voice call stat CallID CID ccVdb Port DSP/Ch Called # Codec Dial-peers 0xC0 13B3 0x49DA073C 50/0/2.0 95002 g729r8 20002/15 1 active call found HQ-RTR(config-gk)#do sh gatek call Total number of active calls = 1. largest hash bucket = 1 GATEKEEPER CALL INFO LocalCallID Age(secs) BW 62-42307 579 33 128(Kbps) ConferenceID CallID SrcCRV A5CF2B43 B52E11E1 81D2B06F 708A730B A5CF2B43 B52E11E1 81D4B06F 708A730B 85 Endpt(s): Alias E.164Addr src EP: BR2-RTR 3002 CallSignalAddr Port RASSignalAddr Port 10.10.110.3 1820 10.10.110.3 62007 Endpt(s): Alias E.164Addr dst EP: gk-trunk_1 95002 CallSignalAddr Port RASSignalAddr Port 10.10.210.10 1720 10.10.210.10 32784 callstate: SEP, DEP, Thank you Krishna. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20120614/7325e5c0/attachment.html -- ___ CCIE_Voice mailing list CCIE_Voice@onlinestudylist.com http://onlinestudylist.com/mailman/listinfo/ccie_voice End of CCIE_Voice Digest, Vol 76, Issue 42 ** This email and any files transmitted with it are confidential and are intended for the sole use of the individual to whom they are addressed. Black Box Corporation reserves the right to scan all e-mail traffic for restricted content and to monitor all e-mail in general. If you are not the intended recipient or you have received this email in error, any use, dissemination or forwarding of this email is strictly prohibited. If you have received this email in error, please notify the sender by replying to this email. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Gatekeeper acting weird for codec
Hi folks, I configured the gatekeeper on the Hq router, and when i call from hq to br2(cme) the call set up shows as 16 kbps, but whereas from cme to hq it shows as 128 kbps but the actual call is connected with g729. Even after the call got connected, it stills shows as 128 in the show gatekeeper call. here is the output status: i put the gk-trunk in the hq region as well. any help is much appreciated on this matter. BR2-RTR(config)#do sh voice call stat CallID CID ccVdb Port DSP/Ch Called # Codec Dial-peers 0xC0 13B3 0x49DA073C 50/0/2.0 95002 g729r8 20002/15 1 active call found HQ-RTR(config-gk)#do sh gatek call Total number of active calls = 1. largest hash bucket = 1 GATEKEEPER CALL INFO LocalCallID Age(secs) BW 62-42307 579 33 128(Kbps) ConferenceID CallID SrcCRV A5CF2B43 B52E11E1 81D2B06F 708A730B A5CF2B43 B52E11E1 81D4B06F 708A730B 85 Endpt(s): Alias E.164Addr src EP: BR2-RTR 3002 CallSignalAddr Port RASSignalAddr Port 10.10.110.3 1820 10.10.110.3 62007 Endpt(s): Alias E.164Addr dst EP: gk-trunk_1 95002 CallSignalAddr Port RASSignalAddr Port 10.10.210.10 1720 10.10.210.10 32784 callstate: SEP, DEP, Thank you Krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] DTMF from BR1 phone to CUC via SIP trunk notrecognized properly
Dan, A small correction to your statement..rfc2833 is out of band mechanism mostly, and moreover it doesn't use audio channel, infact it uses rtp header to relay the dtmf message with a payload identifier. thank you Krishna.. From: Dan Quinlan (daquinla) daqui...@cisco.com To: Tapan Gautam (tgautam) tgau...@cisco.com Cc: ccie_voice@onlinestudylist.com Sent: Monday, June 11, 2012 11:57 PM Subject: Re: [OSL | CCIE_Voice] DTMF from BR1 phone to CUC via SIP trunk notrecognized properly cRTP mangles in-band (audio) DTMF. If I understand correctly, you are SIP-integrated between CUC and UCM. You need to be OOB only for DTMF (not rfc2833). Rfc2833 is an in-band (audio channel) mechanism. You need OOB (signaling channel) for DTMF to function when cRTP is used. DQ d...@cisco.com Sent from my iPhone On Jun 11, 2012, at 11:33 PM, Tapan Gautam (tgautam) tgau...@cisco.com wrote: Hey Guys, When I call CUC pilot from BR1 phone, the dtmf tones are not recognized properly by CUC, i.e. BR1 phone cannot login to mailbox or select any other option via DTMF. If I remove crtp, everything works fine. Topology: SCCP phone(BR1 site) à g729r8 with crtp à CUCM à SIP trunk(with OOB and RFC2833 as dtmf options) à CUC Things I have tried so far, 1) All dtmf options in SIP trunk. 2) Enabled mtp option 3) In CUC, changed codec type to just g711u, just g729 and both(which is the default). I found other posts on this issue but none of them has the solution. Thanks, Tapan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] no trace found but calls get routed br1-Hq
Hi folks, I couldn't understand the call flow between HQ and BR1 which are provisioned/registed in the cucm. here is the detail structure: HQ-phone1 -5002 css-hq-international pt-pt-internal BR1-phone1-1002 css-br1-ld pt-pt-internal Both phones are residing in the partition pt-internal, and br1 is a mgcp site and whereas the hq is the h323 site. when i call 1002 from 5002 or vice versa the call works fine, but when i enable deb isdn q931 or deb voip dialp, i dont see anything. Whereas when i enable RSVP based CAC, i can see the traces with the show sccp connections. could any one help me out how the calls are working in between these two. is it because the phones are registered to cucm, but logically in a different device pool and therefore it routes directly on cucm your help is much appreciated. Thank you. Krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] no trace found but calls get routed br1-Hq
Thank you Dan for providing me the detail info. I assumed the same but not sure with my hypothesis. I am wondering if this is the case, then wan qos will not be able to do much isn't it... for instance yesterday i configured llq-cbfwq with bandwidth of 28 for rtp traffic between hq and br1. And, when i called the 5002 from 1002 the call went thru, and this call put on hold and placed another call and it works fine as well. So, from this analysis can i come to conclusion that only location based cac, or rsvp cac can only the number of calls between these two sites??? Thank you Krishna. From: Dan Quinlan (daquinla) daqui...@cisco.com To: Krishna vinayak_...@yahoo.com Cc: ccie_voice@onlinestudylist.com Sent: Monday, June 11, 2012 2:03 PM Subject: Re: [OSL | CCIE_Voice] no trace found but calls get routed br1-Hq You answered your own question. Both DNs are registered to CUCM and are in partitions that the other's CSS can see. The signaling is between each phone and UCM. The media is built directly phone to phone. If CAC failed the call setup, then AAR could be invoked to use the PSTN. Since CAC allows the call, the gateways aren't involved in the call (other than providing IP network connectivity.) DQ d...@cisco.com Sent from my iPhone On Jun 11, 2012, at 2:52 PM, Krishna vinayak_...@yahoo.com wrote: Hi folks, I couldn't understand the call flow between HQ and BR1 which are provisioned/registed in the cucm. here is the detail structure: HQ-phone1 -5002 css-hq-international pt-pt-internal BR1-phone1-1002 css-br1-ld pt-pt-internal Both phones are residing in the partition pt-internal, and br1 is a mgcp site and whereas the hq is the h323 site. when i call 1002 from 5002 or vice versa the call works fine, but when i enable deb isdn q931 or deb voip dialp, i dont see anything. Whereas when i enable RSVP based CAC, i can see the traces with the show sccp connections. could any one help me out how the calls are working in between these two. is it because the phones are registered to cucm, but logically in a different device pool and therefore it routes directly on cucm your help is much appreciated. Thank you. Krishna. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] no output on interface for qos
hi folks, here is my cbwfq config. After i applied it on the interface dlci it shows nothing in the show policy-map interface.. why this is happening?? your help is much appreciated. BR2-RTR#sh run | s class-map|policy-map|map-class class-map match-any VOIP-CONTROL match protocol skinny match protocol mgcp match protocol sip match protocol rsvp match protocol h323 class-map match-any VOIP-RTP match protocol rtp audio policy-map PVOIP-CONTROL-RTP class VOIP-CONTROL set dscp cs3 bandwidth 66 class VOIP-RTP set ip dscp ef bandwidth 112 class class-default police rate percent 65 exceed-action set-dscp-transmit 0 fair-queue map-class frame-relay LINKPMAP-VOIP-RTP frame-relay cir 729600 frame-relay bc 7296 frame-relay be 0 frame-relay mincir 729600 service-policy output PVOIP-CONTROL-RTP BR2-RTR#sh policy-map interface BR2-RTR# Thank you Krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] no output on interface for qos
plz ignore my last query,... when i implemented the command frame-relay traffic shapping on the physical interface, it showed the output From: Krishna vinayak_...@yahoo.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Sunday, June 10, 2012 6:45 PM Subject: Re: no output on interface for qos hi folks, here is my cbwfq config. After i applied it on the interface dlci it shows nothing in the show policy-map interface.. why this is happening?? your help is much appreciated. BR2-RTR#sh run | s class-map|policy-map|map-class class-map match-any VOIP-CONTROL match protocol skinny match protocol mgcp match protocol sip match protocol rsvp match protocol h323 class-map match-any VOIP-RTP match protocol rtp audio policy-map PVOIP-CONTROL-RTP class VOIP-CONTROL set dscp cs3 bandwidth 66 class VOIP-RTP set ip dscp ef bandwidth 112 class class-default police rate percent 65 exceed-action set-dscp-transmit 0 fair-queue map-class frame-relay LINKPMAP-VOIP-RTP frame-relay cir 729600 frame-relay bc 7296 frame-relay be 0 frame-relay mincir 729600 service-policy output PVOIP-CONTROL-RTP BR2-RTR#sh policy-map interface BR2-RTR# Thank you Krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Qos class class-default will route the voice rtp traffic???
Hi folks, I configured the below LLQ-CBWFQ on the HQ and BR1, and when i made the test calls i.e. two calls made from br1 -hq, the calls went through. But in the configuration the rtp allowed for only one g729 call, so will the class class-default taking this traffic??? Any help in giving clarification is much appreciated. And, also i have a question about mismatch of bandwidth values of sites, by default what values will it take. For example, hq is configured for 2 calls for rtp, br1 configured for 1 call for rtp across the frame-relay, so in this case which one wins??? HQ-RTR(config-pmap-c)# do sh run | s policy-map|class-map|map-class class-map match-any VOIP-CONTROL match protocol skinny match protocol mgcp match protocol sip match protocol rsvp match protocol h323 class-map match-any VOIP-RTP match protocol rtp audio policy-map LLQ-HQ-BR1 class VOIP-CONTROL set dscp cs3 bandwidth 10 class VOIP-RTP set ip dscp ef bandwidth 28 class class-default police rate percent 62 exceed-action set-dscp-transmit 0 fair-queue map-class frame-relay LINKPMAP-VOIP-RTP-BR1 frame-relay cir 146 frame-relay bc 14600 frame-relay be 0 frame-relay mincir 146 service-policy output LLQ-HQ-BR1 Thank you Krishna. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] icd service not showing up
hi folks, http://10.10.210.5:6293/ipphone/jsp/sciphonexml/IPAgentInitial.jsp is the one i created under phone services, but when i want to subsrcibe this service to device i.e.phone, its not showing up. Is there any service activation required to make this work?? your help is greatly appreciated. thank you. Krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] icd service not showing up
Ashwani, you're right. when i uncheck the Enterprise subcription i can see it . thanks for your help dude.. Thank you. Regards, krishna. From: Ashwani ash_r...@hotmail.com To: 'Krishna' vinayak_...@yahoo.com Sent: Saturday, June 9, 2012 6:58 PM Subject: RE: [OSL | CCIE_Voice] icd service not showing up Make you have not checked “Enterprise Subscription” , that will cause Phone services not to show up when you want subscribe/unsubscribe them from phone. From:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Krishna Sent: Saturday, June 09, 2012 5:00 PM To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] icd service not showing up hi folks, http://10.10.210.5:6293/ipphone/jsp/sciphonexml/IPAgentInitial.jsp is the one i created under phone services, but when i want to subsrcibe this service to device i.e.phone, its not showing up. Is there any service activation required to make this work?? your help is greatly appreciated. thank you. Krishna. No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.2178 / Virus Database: 2433/5059 - Release Date: 06/09/12___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Proctor Labs issue?
i have the same issue as well... even most disgusting thing is i cannot access web gui of the servers, and therefore i m using software vpn for accessing the web gui for the servers. thank you krishna. From: Dan Quinlan (daquinla) daqui...@cisco.com To: ccie_voice@onlinestudylist.com Sent: Friday, June 8, 2012 12:51 PM Subject: [OSL | CCIE_Voice] Proctor Labs issue? All, The past two times I've used ProctorLabs for rack time, I've seen my local (local to me) IP Phones deregister / reregister every few minutes and they usually don't come all the way back (DN's missing, etc). I haven't changed my hardware VPN connectivity at all. Has anyone else seen issues lately? Tia DQ d...@cisco.com Sent from my iPhone ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] web login info for ucx/uccx
hi folks, when i copied the standard appadmin and saved it as ucadmin and password info, i tried to login using these credentials into ucx but no luck. isnt this is possible the way i m doing it now??? thank you.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] cannot conference on CME
Dan, you're right we need a dn that needs for conferencing. without this dn, it didnt work. now my concern is: does this conference can only merge 8 active calls or is it capable of taking more calls,since it is a otcal dn??? And, also can you advice how to set up a conference number on this router, so that users can dial into this number and put the passcode for conference into this router. your help is much appreciated. Thank you. From: Dan Quinlan (daquinla) daqui...@cisco.com To: Krishna vinayak_...@yahoo.com Cc: ccie_voice@onlinestudylist.com Sent: Monday, June 4, 2012 5:53 PM Subject: Re: [OSL | CCIE_Voice] cannot conference on CME You need an ephone-dn for conferencing - it does not need to be a dialable number. Add something like: ephone-dn 10 octo number a01 Conference ad-hoc Also, under telephony-service do conference hardware DQ d...@cisco.com Sent from my iPhone On Jun 4, 2012, at 3:52 PM, Krishna vinayak_...@yahoo.com wrote: Hi folks, i configured the transcoder and conference resources on the CME, but couldn't make them to work. When i want to conference the line, it says cannot complete the conference. here is my config: Did i miss any configuration part in this below config sccp local FastEthernet0/0 sccp ccm 10.50.5.1 identifier 1 version 7.0 sccp ! sccp ccm group 1 bind interface FastEthernet0/0 associate ccm 1 priority 1 associate profile 1 register Conference associate profile 2 register mtp(mac-address of sourceinterface) keepalive retries 5 switchover method immediate switchback method immediate switchback interval 5 ! dspfarm profile 2 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 24 associate application SCCP ! dspfarm profile 1 conference codec g711ulaw codec g729r8 codec g729br8 codec g729abr8 codec g729ar8 maximum sessions 4 associate application SCCP here is the status of sccp: SCCP Admin State: UP Gateway Local Interface: FastEthernet0/0 IPv4 Address: 10.50.5.1 Port Number: 2000 IP Precedence: 5 User Masked Codec list: None Call Manager: 10.50.5.1, Port Number: 2000 Priority: N/A, Version: 7.0, Identifier: 1 Trustpoint: N/A Transcoding Oper State: ACTIVE - Cause Code: NONE Active Call Manager: 10.50.5.1, Port Number: 2000 TCP Link Status: CONNECTED, Profile Identifier: 2 Reported Max Streams: 48, Reported Max OOS Streams: 0 Supported Codec: g711ulaw, Maximum Packetization Period: 30 Supported Codec: g711alaw, Maximum Packetization Period: 30 Supported Codec: g729ar8, Maximum Packetization Period: 60 Supported Codec: g729abr8, Maximum Packetization Period: 60 Supported Codec: g729r8, Maximum Packetization Period: 60 Supported Codec: g729br8, Maximum Packetization Period: 60 Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30 Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30 Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30 Conferencing Oper State: ACTIVE - Cause Code: NONE Active Call Manager: 10.50.5.1, Port Number: 2000 TCP Link Status: CONNECTED, Profile Identifier: 1 Reported Max Streams: 32, Reported Max OOS Streams: 0 Supported Codec: g711ulaw, Maximum Packetization Period: 30 Supported Codec: g729r8, Maximum Packetization Period: 60 Supported Codec: g729br8, Maximum Packetization Period: 60 Supported Codec: g729abr8, Maximum Packetization Period: 60 Supported Codec: g729ar8, Maximum Packetization Period: 60 Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30 Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30 Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] cannot conference on CME
Hi folks, i configured the transcoder and conference resources on the CME, but couldn't make them to work. When i want to conference the line, it says cannot complete the conference. here is my config: Did i miss any configuration part in this below config sccp local FastEthernet0/0 sccp ccm 10.50.5.1 identifier 1 version 7.0 sccp ! sccp ccm group 1 bind interface FastEthernet0/0 associate ccm 1 priority 1 associate profile 1 register Conference associate profile 2 register mtp(mac-address of sourceinterface) keepalive retries 5 switchover method immediate switchback method immediate switchback interval 5 ! dspfarm profile 2 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 24 associate application SCCP ! dspfarm profile 1 conference codec g711ulaw codec g729r8 codec g729br8 codec g729abr8 codec g729ar8 maximum sessions 4 associate application SCCP here is the status of sccp: SCCP Admin State: UP Gateway Local Interface: FastEthernet0/0 IPv4 Address: 10.50.5.1 Port Number: 2000 IP Precedence: 5 User Masked Codec list: None Call Manager: 10.50.5.1, Port Number: 2000 Priority: N/A, Version: 7.0, Identifier: 1 Trustpoint: N/A Transcoding Oper State: ACTIVE - Cause Code: NONE Active Call Manager: 10.50.5.1, Port Number: 2000 TCP Link Status: CONNECTED, Profile Identifier: 2 Reported Max Streams: 48, Reported Max OOS Streams: 0 Supported Codec: g711ulaw, Maximum Packetization Period: 30 Supported Codec: g711alaw, Maximum Packetization Period: 30 Supported Codec: g729ar8, Maximum Packetization Period: 60 Supported Codec: g729abr8, Maximum Packetization Period: 60 Supported Codec: g729r8, Maximum Packetization Period: 60 Supported Codec: g729br8, Maximum Packetization Period: 60 Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30 Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30 Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30 Conferencing Oper State: ACTIVE - Cause Code: NONE Active Call Manager: 10.50.5.1, Port Number: 2000 TCP Link Status: CONNECTED, Profile Identifier: 1 Reported Max Streams: 32, Reported Max OOS Streams: 0 Supported Codec: g711ulaw, Maximum Packetization Period: 30 Supported Codec: g729r8, Maximum Packetization Period: 60 Supported Codec: g729br8, Maximum Packetization Period: 60 Supported Codec: g729abr8, Maximum Packetization Period: 60 Supported Codec: g729ar8, Maximum Packetization Period: 60 Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30 Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30 Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CUPS not covered in depth
hi folks, can anyone advise me which document will clearly explains the cups integration and its logical connectivity??? the vod didn't really covered the concepts or walk through videos with out explaining the depth in much. i feel that Vik didn't put much efforts at the end especially for UCCX and CUPS section. I am completely lost in both these sections, and i am trying to learn these two in order to complete the vol1 workbook completely. thank you. regards, Krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] hunt group via AA not working
Thanks Mohammed. I think Vik has to change his volume 1 solution guide as well accordingly taking this fact into consideration.From: Mohammed Al Baqari baqari.voic...@gmail.com To: 'Justin McIntyre' justin.mcint...@blackbox.com Cc: ccie_voice@onlinestudylist.com Sent: Thursday, May 24, 2012 2:21 AM Subject: Re: [OSL | CCIE_Voice] hunt group via AA not working Here you go. Regards,Mohammed Al Baqari -Original Message-From: Justin McIntyre [mailto:justin.mcint...@blackbox.com] Sent: Thursday, May 24, 2012 1:05 AMTo: Mohd BaqariCc: ccie_voice@onlinestudylist.comSubject: RE: [OSL | CCIE_Voice] hunt group via AA not working There it is . I couldn't remember which one it supported. Thanks, Justin McIntyreEngineerMutual Telecom Services Inc.a wholly-owned subsidiary of Black Box Corp.COMM: (434)-946-1562DSN: (312)-237-1562CELL: (540)-312-9391FAX: (434)-946-1510-Original Message-From: Mohd Baqari [mailto:baqari.voic...@gmail.com]Sent: Wednesday, May 23, 2012 3:03 PMTo: Justin McIntyreCc: ccie_voice@onlinestudylist.comSubject: Re: [OSL | CCIE_Voice] hunt group via AA not working Plz share the full config including hunt groups. I remeber that bacd supports ephone hunt but not voice hint groups. Regards,Mohammed Al Baqari Sent from my iPhone On May 23, 2012, at 4:06 PM, Justin McIntyre justin.mcint...@blackbox.com wrote: Is there any way we could see the rest of your config? Where is the Hunt group configured? Are the phones running in CME,UCM? I have seen it once and it was due to the way I configured the hunt group in CME. I used hunt-group instead of Voice hunt-group and I think that's what caused it to break going through the BACD AA application. Just a thought. Thanks, JustinMessage: 4 Date: Tue, 22 May 2012 20:28:54 -0700 (PDT) From: Krishna vinayak_...@yahoo.com To: "ccie_voice@onlinestudylist.com" ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] hunt group via AA not working Message-ID: 1337743734.93075.yahoomail...@web46001.mail.sp1.yahoo.com Content-Type: text/plain; charset="iso-8859-1" Folks, it is so strange that when i call the hunt group number 3210 from pstn, both sip and sccp phone rings. But with AA on cme, only cisco phone rings but not both. even i verified with the config, and i see hunt group as the right option when user presses the digit 2. does anyone know why this is happening only for AA??? here is the config: application ?service queue flash:bacdprompts/app-b-acd-2.1.2.2.tcl ? param number-of-hunt-grps 2 ? param aa-hunt2 3210 ? param aa-hunt10 3006 ? param queue-len 15 ? param queue-manager-debugs 1 service aa flash:bacdprompts/app-b-acd-aa-2.1.2.2.tcl ? paramspace english index 1 ? paramspace english language en ? paramspace english location flash:bacdprompts/ ? param service-name queue ? param handoff-string aa ? param aa-pilot 3500 ? param welcome-prompt _bacd_welcome.au ? param number-of-hunt-grps 2 ? param second-greeting-time 60 ? param call-retry-timer 15 ? param max-time-call-retry 700 ? param max-time-vm-retry 2 ? param voice-mail 3001 -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20120522/47dc2af6/attachment.html -- __ This email and any files transmitted with it are confidential and are intended for the sole use of the individual to whom they are addressed. Black Box Corporation reserves the right to scan all e-mail traffic for restricted content and to monitor all e-mail in general. If you are not the intended recipient or you have received this email in error, any use, dissemination or forwarding of this email is strictly prohibited. If you have received this email in error, please notify the sender by replying to this email. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -No virus found in this message.Checked by AVG - www.avg.comVersion: 2012.0.1913 / Virus Database: 2425/5017 - Release Date: 05/23/12___For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comAre you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] hunt group via AA not working
Folks, it is so strange that when i call the hunt group number 3210 from pstn, both sip and sccp phone rings. But with AA on cme, only cisco phone rings but not both. even i verified with the config, and i see hunt group as the right option when user presses the digit 2. does anyone know why this is happening only for AA??? here is the config: application service queue flash:bacdprompts/app-b-acd-2.1.2.2.tcl param number-of-hunt-grps 2 param aa-hunt2 3210 param aa-hunt10 3006 param queue-len 15 param queue-manager-debugs 1 service aa flash:bacdprompts/app-b-acd-aa-2.1.2.2.tcl paramspace english index 1 paramspace english language en paramspace english location flash:bacdprompts/ param service-name queue param handoff-string aa param aa-pilot 3500 param welcome-prompt _bacd_welcome.au param number-of-hunt-grps 2 param second-greeting-time 60 param call-retry-timer 15 param max-time-call-retry 700 param max-time-vm-retry 2 param voice-mail 3001___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] BAT tool for phone install
hi folks, where can i find the bat tool for 7961 phones as it explicitly defined in the workbook volume 1. Everytime i am deleting the 7962 phones, and adding my 7961 phones which obviously wasting a little time when i load the configs everytime for a new task. could you please help me out where is the bat tool for 7961, your help is much appreciated. Thank you. Regards, Krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] pots dial-peer showing down but calls are connecting
Hello... did anyone experience the scenario which i experienced for gateway lab config. ?? basically the dial-p v 1 pots is for pots incoming dial-peer, and when i issued the command sh dial-p voi summ, the pots dial-p 1 is showing as out statust down, but strange thing is that calls are connecting through the same dial-peer. AD PRE PASS OUT TAG TYPE MIN OPER PREFIX DEST-PATTERN FER THRU SESS-TARGET STAT PORT 20002 pots up up 3002$ 0 50/0/2 20003 pots up up 3001$ 0 50/0/1 20005 pots up down 0 50/0/3 999 pots up up 999 0 up 0/0/0:15 101 voip up up 0 syst 40002 voip up up 3005 0 syst ipv4:10.10.202.50:50 40001 voip up up 3006 0 syst ipv4:192.168.10.13:5 1 pots up up 0 down 0/0/0:15 BR2-RTR(config-ephone)#do sh voice call stat CallID CID ccVdb Port DSP/Ch Called # Codec Dial-peers 0x129 121E 0x498213FC 0/0/0:15.1 0/5:1 3002 g729r8 1/20002 0x12A 121E 0x49821034 50/0/2.0 *3002 g729r8 20002/1 1 active call found please advise me on this perspective. Thank you in advance.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] SIP phone upgrading firmware everytime with a restart /reset
Hello Friends, I was little bit dazed about the SIP ip phone that is registered via hardware vpn. The issue when ever i issue restart/reset, the sip phone at my desk starts the process with upgrading the phone and then registers, whereas the vrack one registers normally immediately. Can anyone advice what would causing this issue?? BR2-RTR(config-register-global)#restart BR2-RTR(config-register-global)# May 2 03:45:44.188: VOICE REGISTER POOL-1 has unregistered. Name:SEP0018B9788DCF IP:10.10.202.50 DeviceType:Phone May 2 03:45:44.500: VOICE REGISTER POOL-2 has unregistered. Name:SEP00270DBFC491 IP:192.168.10.11 DeviceType:Phone BR2-RTR(config-register-global)# May 2 03:45:47.752: VOICE REGISTER POOL-1 has registered. Name:SEP0018B9788DCF IP:10.10.202.50 DeviceType:Phone May 2 03:54:57.105: TFTP: Looking for CTLSEP00270DBFC491.tlv May 2 03:54:57.681: TFTP: Looking for SEP00270DBFC491.cnf.xml May 2 03:54:57.685: TFTP: Opened flash:/SEP00270DBFC491.cnf.xml, fd 7, size 2711 for process 341 BR2-RTR(config)# May 2 03:54:58.265: TFTP: Finished flash:/SEP00270DBFC491.cnf.xml, time 00:00:00 for process 341 BR2-RTR(config)# May 2 03:55:09.305: TFTP: Looking for SIP41.8-4-3S.loads May 2 03:55:09.309: TFTP: Opened flash:PHONE/7941-7961/SIP41.8-4-3S.loads, fd 7, size 638 for process 341 May 2 03:55:09.497: TFTP: Finished flash:PHONE/7941-7961/SIP41.8-4-3S.loads, time 00:00:00 for process 341 BR2-RTR(config)# May 2 03:55:10.449: TFTP: Looking for jar41sip.8-4-2-38.sbn May 2 03:55:10.457: TFTP: Opened flash:jar41sip.8-4-2-38.sbn, fd 7, size 448988 for process 341___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] firmware for SCCP/SIP phones
Hello ALL, Can anyone advice me what version of sip firmware or sccp firmware should be given in tftp-server command in the router config.??? for instance: the version for sccp are : PHONE/7940-7960/P003-08-6-00.bin PHONE/7940-7960/P003-08-9-00.bin PHONE/7940-7960/P00308000500.bin For SIP: PHONE/7940-7960/P0S3-08-6-00.loads PHONE/7940-7960/P0S3-08-9-00.loads Can we update the phones with whatever version that can be chosen from flash or is there any ground work has to be done to make sure that this is the right file for this sip/sccp phone. Any help on this matter is appreciated. Thank you.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] sip phone registered with sccp file
Hi Guys, When I am doing my Lab 3B, i encountered a strange problem for sip phone. Isn't the sip phones required firmware that starts with P0S??? vrack SIP phone never tried to register though all P0S firmware commands available for the phone, but the registration is still rejected. When i did shut and no shut , on the interface i found that it is looking for a file P003-08-6-00.sbn, and when i included this file in the tftp-server, my sip ip phone registered normally. can anyone clarify me that P00 files do needed for sip phones as well??? BR2-RTR(config-if)# May 2 06:47:24.404: TFTP: Looking for P003-08-6-00.sbn thank you. Regards, Krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] glitch in the lab1B
Hello.. Those who are doing the vol1 workbook, there is a glitch in the lab1B: We define Pub as dhcp server address, whereas the solutions guide refers to work to Sub in troubleshooting the dhcp problem. I finally realized that dhcp server is enabled on Pub, not on Sub. So, I changed the helper address on HQ-RTR to Publisher IP rather subscriber to fix this issue. thank you.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] connectivity via Hardware vpn
Hi... I am running in to an issue where my hardware IP phones are unable to register with call manager. I am using hardware vpn set up for connecting IP phones. here is my set up: IP phones -- 2924xl switch--- 2811 router--- Internet. My pc could able to access cucm servers, where IP phones are not getting registered with cucm. Can anyone assist me on this perspective??? I cannot proceed any further labs with out this working. Pretty much i got stuck for the whole day with this issue, and still it is not yet resolved. Any help would be much appreciated and commendable. Note: I used the exactly the config of 'Router switch' which is in the proctor labs. Thank you in advance. regards, Krishna Koilada.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] NTP Set up lab 1A
Can anyone advise me where the IP address 10.10.200.2 came from??? the question is to configure the PSTN-WAN router to synchronize with cucm. I am little bit confused, can anyone help me in this matter.. thanks in advance. Regards, Krishna.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] High Availabilty using route group
All, I;d like to receive feedback and confirmation about implementing the HA system, and this is about concept of design and implementation as well. please, advise me on this perspective. Here is the existing call flow : gateway(site)--cucm---Route Group cubes--Service provider (sip trunk) But, I want this call flow to be redundant by adding the second service provider as well. My question: Is it possible using Route Group feature to send only calls to SP2 in case SP 1 down,, if not then how can i achieve HA with 2 SP using RG's??? gateway(site)--cucm---Route Group cubes--Service provider 1 (sip trunk) --- Service provider 2 (sip trunk) Your help is much appreciated. thank you. Regards, Krishna Koilada. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] respond 200OK for sip reinvite on CUBE
Hi All, I did my best in resolving the sip reinvite issue for AS5400XM cube, but couldn't find the solution for Version 12.4.24 -T5. The issue is the carrier sending the sip reinvite intermittently even min-se is set to 1800(30 minutes) for the established call, and I tried to block that sip reinvite at CUBE level, and to respond with 200ok to keep the session alive, but due to no response from the CUBE, the session is being terminated by the vendor/carrier by sending the bye . I am figuring out a way to fix this problem rather updating the IOS to 15 version, and implementing the mid-call signalling block under voice services. Any help on this greatly appreciated and admirable. Thank you.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
