[OSL | CCIE_Voice] cbarge softkey display on phone
Hi All , Need clarification on Cbarge behavior. here is the config Cbarge for brach phone shared line on both phone device pool of both phone should have hardware conf bridge privacy should be disabled Single button barge should be set to Cbarge built in bridge should be disabled soft key template where remote in use has two state Cbarge ,New call is applied to both phone when there is active call on phone 1 and phone 2 is pressing the shared button , conference is being established without offering soft key options “new call , cBarge” to user on phone display ! is this expected ?? thx , Rahul ___ 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] cbarge softkey display on phone
you wrote Single button barge should be set to Cbarge this allows you to barge using button. no softkey will be offered. this is expected behaviour. -adil On Tue, May 31, 2011 at 5:26 PM, Rahul Kapor rahul.kapo...@gmail.comwrote: Hi All , Need clarification on Cbarge behavior. here is the config Cbarge for brach phone shared line on both phone device pool of both phone should have hardware conf bridge privacy should be disabled Single button barge should be set to Cbarge built in bridge should be disabled soft key template where remote in use has two state Cbarge ,New call is applied to both phone when there is active call on phone 1 and phone 2 is pressing the shared button , conference is being established without offering soft key options “new call , cBarge” to user on phone display ! is this expected ?? thx , Rahul ___ 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 -- .. . . _7___|___|_|_|adil.sha...@gmail.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] Multicast MoH for IP Phones
Hi, I have configured Mulitcast MoH for Branch office using ccm-manager music-on-hold , when there is PSTN call to Branch and the Branch user places him on hold , I can see from the show ccm-manager music-on-hold as the Multicast is being used, but the same does not work when the Central site IP Phone is put on hold. Unicast moh is heard from Central site IP Phones. Not sure if I am missing anything on the configuration part. Warm Regards, Vinay Kumar ___ 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] Multicast MoH for IP Phones
Multicast MOH x.x.x.x port x route subnetgw Subnetgw = Ur voice vlan gw ip Sent from my iPhone Pls pardon my fat fingers. On May 31, 2011, at 5:48 PM, Vinay Kumar6 vinayjaisw...@in.ibm.com wrote: Hi, I have configured Mulitcast MoH for Branch office using ccm-manager music-on-hold , when there is PSTN call to Branch and the Branch user places him on hold , I can see from the show ccm-manager music-on-hold as the Multicast is being used, but the same does not work when the Central site IP Phone is put on hold. Unicast moh is heard from Central site IP Phones. Not sure if I am missing anything on the configuration part. Warm Regards, Vinay Kumar ___ 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] Multicast MoH for IP Phones
I have this command in my config but still for the IP Phones registered at Central site it plays the MOH from CUCM. Warm Regards, Vinay Kumar From: Ki Wi kiwi.vo...@gmail.com To: Vinay Kumar6/India/IBM@IBMIN Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: 05/31/2011 04:54 PM Subject: Re: [OSL | CCIE_Voice] Multicast MoH for IP Phones Multicast MOH x.x.x.x port x route subnetgw Subnetgw = Ur voice vlan gw ip Sent from my iPhone Pls pardon my fat fingers. On May 31, 2011, at 5:48 PM, Vinay Kumar6 vinayjaisw...@in.ibm.com wrote: Hi, I have configured Mulitcast MoH for Branch office using ccm-manager music-on-hold , when there is PSTN call to Branch and the Branch user places him on hold , I can see from the show ccm-manager music-on-hold as the Multicast is being used, but the same does not work when the Central site IP Phone is put on hold. Unicast moh is heard from Central site IP Phones. Not sure if I am missing anything on the configuration part. Warm Regards, Vinay Kumar ___ 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] Multicast MoH for IP Phones
Ensure you create MOH dp. You can do debug ephone MOH and use rtmt to see whether mmoh is being triggered or not Sent from my iPhone Pls pardon my fat fingers. On May 31, 2011, at 8:23 PM, Vinay Kumar6 vinayjaisw...@in.ibm.com wrote: I have this command in my config but still for the IP Phones registered at Central site it plays the MOH from CUCM. Warm Regards, Vinay Kumar From: Ki Wi kiwi.vo...@gmail.com To: Vinay Kumar6/India/IBM@IBMIN Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: 05/31/2011 04:54 PM Subject: Re: [OSL | CCIE_Voice] Multicast MoH for IP Phones Multicast MOH x.x.x.x port x route subnetgw Subnetgw = Ur voice vlan gw ip Sent from my iPhone Pls pardon my fat fingers. On May 31, 2011, at 5:48 PM, Vinay Kumar6 vinayjaisw...@in.ibm.com wrote: Hi, I have configured Mulitcast MoH for Branch office using ccm-manager music-on-hold , when there is PSTN call to Branch and the Branch user places him on hold , I can see from the show ccm-manager music-on-hold as the Multicast is being used, but the same does not work when the Central site IP Phone is put on hold. Unicast moh is heard from Central site IP Phones. Not sure if I am missing anything on the configuration part. Warm Regards, Vinay Kumar ___ 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] Multicast MoH for IP Phones
Thanks a lot Ki Wi, Multicast moh is working now for calls between the Branch phone and the Central phone. The calls which are passing through the gatekeeper are getting ToH when the CUCME phones are put on hold, if the HQ phones are put on hold the Multicast is played from the flash of the CUCME. Warm Regards, Vinay Kumar From: Ki Wi kiwi.vo...@gmail.com To: Vinay Kumar6/India/IBM@IBMIN Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: 05/31/2011 06:25 PM Subject: Re: [OSL | CCIE_Voice] Multicast MoH for IP Phones Ensure you create MOH dp. You can do debug ephone MOH and use rtmt to see whether mmoh is being triggered or not Sent from my iPhone Pls pardon my fat fingers. On May 31, 2011, at 8:23 PM, Vinay Kumar6 vinayjaisw...@in.ibm.com wrote: I have this command in my config but still for the IP Phones registered at Central site it plays the MOH from CUCM. Warm Regards, Vinay Kumar From: Ki Wi kiwi.vo...@gmail.com To: Vinay Kumar6/India/IBM@IBMIN Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: 05/31/2011 04:54 PM Subject: Re: [OSL | CCIE_Voice] Multicast MoH for IP Phones Multicast MOH x.x.x.x port x route subnetgw Subnetgw = Ur voice vlan gw ip Sent from my iPhone Pls pardon my fat fingers. On May 31, 2011, at 5:48 PM, Vinay Kumar6 vinayjaisw...@in.ibm.com wrote: Hi, I have configured Mulitcast MoH for Branch office using ccm-manager music-on-hold , when there is PSTN call to Branch and the Branch user places him on hold , I can see from the show ccm-manager music-on-hold as the Multicast is being used, but the same does not work when the Central site IP Phone is put on hold. Unicast moh is heard from Central site IP Phones. Not sure if I am missing anything on the configuration part. Warm Regards, Vinay Kumar ___ 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] SW VPN Vs HW VPN during the Proctor Lab Session
Hi, Currently i am using software vpn client to connect to my rack session. Using IP Blue clients, IP Communicator etc.. to practice the work books. How ever i am confused whether i may need hardware phones to test out the specific features like + dialing, etc..moving forward. Is it a must that we will require hardware phones during the proctor lab session to test all the features? Or only the soft phones will help us till we succeed the lab? Please advice. Ilham ___ 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] SW VPN Vs HW VPN during the Proctor Lab Session
is not a must with soft clients sometimes you assume your config is working, so you never troubleshoot because you just can with hardware phones you can test most off the features if not all of them 2011/5/31 Ilham Jabir ciscotodi...@gmail.com Hi, Currently i am using software vpn client to connect to my rack session. Using IP Blue clients, IP Communicator etc.. to practice the work books. How ever i am confused whether i may need hardware phones to test out the specific features like + dialing, etc..moving forward. Is it a must that we will require hardware phones during the proctor lab session to test all the features? Or only the soft phones will help us till we succeed the lab? Please advice. Ilham ___ 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 on cisco7961 didn't alert
Hi Ki Wi, I've encounter the same issue also, and I solved it by changing the Enterprise Parameters Services URL to IP instead of hostname (Apparently, I miss that part when I reverted my VMware snapshot), remember I saw this solution from OSL discussion before. HTH Cheers, Alex On Tue, May 31, 2011 at 11:25 AM, ShinGei Yong shingei.y...@gmail.comwrote: Frens, If i can recall correctly, that was due to that i missed associate the phone with application user Phone Messenger. You need the phone messenger application user to control the IPPM user. Without the association, the messaging will still work but funny stuff come out, if not wrong Shingei. On Tue, May 31, 2011 at 4:52 AM, Ki Wi kiwi.vo...@gmail.com wrote: Hey, Do you still remember how did you resolve this alert issue? I'm still trying to train myself up in CUPS. Last night, my alert was working, my IPPM login wasn't. Today my IPPM is working but no alert. =( All other components are working. On Sun, Dec 26, 2010 at 12:59 AM, ShinGei Yong shingei.y...@gmail.comwrote: Guys, Pls ignore this mail, has managed to figured out the caused. thanks Shingei. On Sat, Dec 25, 2010 at 4:36 PM, ShinGei Yong shingei.y...@gmail.comwrote: Hi, I've configure the IPPM on cisco 7961 phone, everything works smooth other that the message receive alert. It doesn't ring when there is a mgs come in from CIPC or other IPPM.i've set the audible alert to ON but still got no luck. Another IPPM phone encounter the same issue, so don't think is the phone problem. Any idea? Thanks Shingei. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com 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] AIM-CUE CF card problem
Hi Guys, Hope I can seek a little help here, my AIM-CUE 1GB CF card failed on me just 1 week before my exam! I've getting the error of Not a cisco supported CF. Please use cisco supported CF and reinstall the software. System Halted. Anyone know how to solved this issue? I've try to reinstall CUE using the boothelper, but no luck. Possibly the CF card is gone case. A search on google mentioned Cisco AIM-CUE check on the CF Card sector size, else refuse to work. But the used 1GB CF card was asking half the price of the AIM-CUE module /w 1GB CF itself on ebay :( It is anyway I can used on 3rd party CF card? saw it also certain SANDISK CF might work, but I'm not sure it is still able to find in the market now. Also, I notice the router Memory CF (MEM-CF-1GB) is selling cheaper than AIM-CUE-1GBCF, I wonder will it able to use? Any help will be appreciated. Regards, Alex ___ 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] 3750 QoS
guys i have a question for your my questions is in regards to the shared bandwidth i know it's measured as a weight and whenever you have shape you ignore the value so the q2 in this case the weight is indeed 10 / (10+60+20) = so Q2 has 1/9 of the shared bandwidth right ? and this is how the config looks like interface FastEthernet0/1 switchport access vlan 6 switchport voice vlan 8 srr-queue bandwidth share 10 10 60 20 srr-queue bandwidth shape 10 0 0 0 mls qos trust device cisco-phone mls qos trust cos auto qos voip cisco-phone spanning-tree portfast so if the bandwith needs to be shared on q2 40 %, q3 20%, q4 40% how could i calculate or know what do i need to put on srr-queue bandwidth share command will this be an accurate config srr-queue bandwidth share 0 40 20 40 ?? 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
[OSL | CCIE_Voice] Call Preservation Issue
Hello Experts An interesting problem to solve for which I'm having a hard time figuring out the solution myself. We have a Call Manager node and IP Phones in one location and ingress voice gateways in other location. Both locations are connected over redundant WAN links. When one WAN link fails, it takes upto 5 seconds for the redundant WAN link to kick in. Issue is, if there is an active call between the gateway and the IP Phone, the call simply drops when a WAN issue occurs. Is there a way to preserve the RTP stream on the GW itself that can be resumed once the redundant WAN link takes over? Any input is appreciated! ___ 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] 3750 QoS
Answer your 1st question, its 1/10th. 2nd question is correct. duy ccie #27737 voice tmobile g2 On May 31, 2011 4:19 PM, Cristobal Priego cristobalpri...@gmail.com wrote: guys i have a question for your my questions is in regards to the shared bandwidth i know it's measured as a weight and whenever you have shape you ignore the value so the q2 in this case the weight is indeed 10 / (10+60+20) = so Q2 has 1/9 of the shared bandwidth right ? and this is how the config looks like interface FastEthernet0/1 switchport access vlan 6 switchport voice vlan 8 srr-queue bandwidth share 10 10 60 20 srr-queue bandwidth shape 10 0 0 0 mls qos trust device cisco-phone mls qos trust cos auto qos voip cisco-phone spanning-tree portfast so if the bandwith needs to be shared on q2 40 %, q3 20%, q4 40% how could i calculate or know what do i need to put on srr-queue bandwidth share command will this be an accurate config srr-queue bandwidth share 0 40 20 40 ?? 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
Re: [OSL | CCIE_Voice] AIM-CUE CF card problem
The mem is your ram, CF is your hd. Try the sandisk CF, I'm sure cisco just oem it from either sandisk or another manufacturer. duy ccie #27737 voice tmobile g2 On May 31, 2011 1:22 PM, Alex Goh ncsalex@gmail.com wrote: Hi Guys, Hope I can seek a little help here, my AIM-CUE 1GB CF card failed on me just 1 week before my exam! I've getting the error of Not a cisco supported CF. Please use cisco supported CF and reinstall the software. System Halted. Anyone know how to solved this issue? I've try to reinstall CUE using the boothelper, but no luck. Possibly the CF card is gone case. A search on google mentioned Cisco AIM-CUE check on the CF Card sector size, else refuse to work. But the used 1GB CF card was asking half the price of the AIM-CUE module /w 1GB CF itself on ebay :( It is anyway I can used on 3rd party CF card? saw it also certain SANDISK CF might work, but I'm not sure it is still able to find in the market now. Also, I notice the router Memory CF (MEM-CF-1GB) is selling cheaper than AIM-CUE-1GBCF, I wonder will it able to use? Any help will be appreciated. Regards, Alex ___ 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] 3750 QoS
thank you 2011/5/31 ccieid1ot ccieid...@gmail.com Answer your 1st question, its 1/10th. 2nd question is correct. duy ccie #27737 voice tmobile g2 On May 31, 2011 4:19 PM, Cristobal Priego cristobalpri...@gmail.com wrote: guys i have a question for your my questions is in regards to the shared bandwidth i know it's measured as a weight and whenever you have shape you ignore the value so the q2 in this case the weight is indeed 10 / (10+60+20) = so Q2 has 1/9 of the shared bandwidth right ? and this is how the config looks like interface FastEthernet0/1 switchport access vlan 6 switchport voice vlan 8 srr-queue bandwidth share 10 10 60 20 srr-queue bandwidth shape 10 0 0 0 mls qos trust device cisco-phone mls qos trust cos auto qos voip cisco-phone spanning-tree portfast so if the bandwith needs to be shared on q2 40 %, q3 20%, q4 40% how could i calculate or know what do i need to put on srr-queue bandwidth share command will this be an accurate config srr-queue bandwidth share 0 40 20 40 ?? 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
Re: [OSL | CCIE_Voice] IPPM on cisco7961 didn't alert
For me, everything is fine. I just do everything again from fresh. Same problem. Something must be missing along the way. Is there any IPPM guide online? The one i found from http://docwiki.cisco.com/wiki/Cisco_Unified_Presence%2C_Release_7.x_--_Configuring_Cisco_IP_Phone_Messenger_on_Cisco_Unified_Presenceis pretty useless. On Wed, Jun 1, 2011 at 1:21 AM, Alex Goh ncsalex@gmail.com wrote: Hi Ki Wi, I've encounter the same issue also, and I solved it by changing the Enterprise Parameters Services URL to IP instead of hostname (Apparently, I miss that part when I reverted my VMware snapshot), remember I saw this solution from OSL discussion before. HTH Cheers, Alex On Tue, May 31, 2011 at 11:25 AM, ShinGei Yong shingei.y...@gmail.comwrote: Frens, If i can recall correctly, that was due to that i missed associate the phone with application user Phone Messenger. You need the phone messenger application user to control the IPPM user. Without the association, the messaging will still work but funny stuff come out, if not wrong Shingei. On Tue, May 31, 2011 at 4:52 AM, Ki Wi kiwi.vo...@gmail.com wrote: Hey, Do you still remember how did you resolve this alert issue? I'm still trying to train myself up in CUPS. Last night, my alert was working, my IPPM login wasn't. Today my IPPM is working but no alert. =( All other components are working. On Sun, Dec 26, 2010 at 12:59 AM, ShinGei Yong shingei.y...@gmail.comwrote: Guys, Pls ignore this mail, has managed to figured out the caused. thanks Shingei. On Sat, Dec 25, 2010 at 4:36 PM, ShinGei Yong shingei.y...@gmail.comwrote: Hi, I've configure the IPPM on cisco 7961 phone, everything works smooth other that the message receive alert. It doesn't ring when there is a mgs come in from CIPC or other IPPM.i've set the audible alert to ON but still got no luck. Another IPPM phone encounter the same issue, so don't think is the phone problem. Any idea? Thanks Shingei. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com 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 on cisco7961 didn't alert
Finally found the problem, I didn't give enough right to IPPM user. Seems like it need CCM Super user right instead! On Wed, Jun 1, 2011 at 6:07 AM, Ki Wi kiwi.vo...@gmail.com wrote: For me, everything is fine. I just do everything again from fresh. Same problem. Something must be missing along the way. Is there any IPPM guide online? The one i found from http://docwiki.cisco.com/wiki/Cisco_Unified_Presence%2C_Release_7.x_--_Configuring_Cisco_IP_Phone_Messenger_on_Cisco_Unified_Presenceis pretty useless. On Wed, Jun 1, 2011 at 1:21 AM, Alex Goh ncsalex@gmail.com wrote: Hi Ki Wi, I've encounter the same issue also, and I solved it by changing the Enterprise Parameters Services URL to IP instead of hostname (Apparently, I miss that part when I reverted my VMware snapshot), remember I saw this solution from OSL discussion before. HTH Cheers, Alex On Tue, May 31, 2011 at 11:25 AM, ShinGei Yong shingei.y...@gmail.comwrote: Frens, If i can recall correctly, that was due to that i missed associate the phone with application user Phone Messenger. You need the phone messenger application user to control the IPPM user. Without the association, the messaging will still work but funny stuff come out, if not wrong Shingei. On Tue, May 31, 2011 at 4:52 AM, Ki Wi kiwi.vo...@gmail.com wrote: Hey, Do you still remember how did you resolve this alert issue? I'm still trying to train myself up in CUPS. Last night, my alert was working, my IPPM login wasn't. Today my IPPM is working but no alert. =( All other components are working. On Sun, Dec 26, 2010 at 12:59 AM, ShinGei Yong shingei.y...@gmail.comwrote: Guys, Pls ignore this mail, has managed to figured out the caused. thanks Shingei. On Sat, Dec 25, 2010 at 4:36 PM, ShinGei Yong shingei.y...@gmail.comwrote: Hi, I've configure the IPPM on cisco 7961 phone, everything works smooth other that the message receive alert. It doesn't ring when there is a mgs come in from CIPC or other IPPM.i've set the audible alert to ON but still got no luck. Another IPPM phone encounter the same issue, so don't think is the phone problem. Any idea? Thanks Shingei. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com 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] SW VPN Vs HW VPN during the Proctor Lab Session
So i can re use the 2$$$ to purchase few 100 hrs of proctor lab session instead of buying the partial hardware which i will use during the rack session. Bottom line is that we can survive with only with soft phones proctor labs for practicing the work books prepare for the exam. Thanks Ilham On 01-Jun-2011, at 1:03 AM, Cristobal Priego wrote: is not a must with soft clients sometimes you assume your config is working, so you never troubleshoot because you just can with hardware phones you can test most off the features if not all of them 2011/5/31 Ilham Jabir ciscotodi...@gmail.com Hi, Currently i am using software vpn client to connect to my rack session. Using IP Blue clients, IP Communicator etc.. to practice the work books. How ever i am confused whether i may need hardware phones to test out the specific features like + dialing, etc..moving forward. Is it a must that we will require hardware phones during the proctor lab session to test all the features? Or only the soft phones will help us till we succeed the lab? Please advice. Ilham ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] SW VPN Vs HW VPN during the Proctor Lab Session
yes no problem i know 2 guys that got their ccie voice and they were only using soft phones 2011/5/31 Ilham Jabir ciscotodi...@gmail.com So i can re use the 2$$$ to purchase few 100 hrs of proctor lab session instead of buying the partial hardware which i will use during the rack session. Bottom line is that we can survive with only with soft phones proctor labs for practicing the work books prepare for the exam. Thanks Ilham On 01-Jun-2011, at 1:03 AM, Cristobal Priego wrote: is not a must with soft clients sometimes you assume your config is working, so you never troubleshoot because you just can with hardware phones you can test most off the features if not all of them 2011/5/31 Ilham Jabir ciscotodi...@gmail.com Hi, Currently i am using software vpn client to connect to my rack session. Using IP Blue clients, IP Communicator etc.. to practice the work books. How ever i am confused whether i may need hardware phones to test out the specific features like + dialing, etc..moving forward. Is it a must that we will require hardware phones during the proctor lab session to test all the features? Or only the soft phones will help us till we succeed the lab? Please advice. Ilham ___ 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] SW VPN Vs HW VPN during the Proctor Lab Session
Thanks for the advice Chris. On 01-Jun-2011, at 6:55 AM, Cristobal Priego wrote: yes no problem i know 2 guys that got their ccie voice and they were only using soft phones 2011/5/31 Ilham Jabir ciscotodi...@gmail.com So i can re use the 2$$$ to purchase few 100 hrs of proctor lab session instead of buying the partial hardware which i will use during the rack session. Bottom line is that we can survive with only with soft phones proctor labs for practicing the work books prepare for the exam. Thanks Ilham On 01-Jun-2011, at 1:03 AM, Cristobal Priego wrote: is not a must with soft clients sometimes you assume your config is working, so you never troubleshoot because you just can with hardware phones you can test most off the features if not all of them 2011/5/31 Ilham Jabir ciscotodi...@gmail.com Hi, Currently i am using software vpn client to connect to my rack session. Using IP Blue clients, IP Communicator etc.. to practice the work books. How ever i am confused whether i may need hardware phones to test out the specific features like + dialing, etc..moving forward. Is it a must that we will require hardware phones during the proctor lab session to test all the features? Or only the soft phones will help us till we succeed the lab? Please advice. Ilham ___ 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] Call Preservation Issue
Sorry, should have been more clear. Using H.323. The call preservation feature of H323 applies when the gateway loses connectivity to Call Manager only, so that the RTP stream is preserved to the phone, but here the RTP stream itself is being interrupted. On May 31, 2011, at 6:58 PM, amit batra batraji...@yahoo.com wrote: What type of protocol you are using ? H323 or MGCP ? With MGCP i dont think if there is any way .. H323 can preserve call .. --- On Wed, 6/1/11, CCIE VOICE ccievoicelab.c...@gmail.com wrote: From: CCIE VOICE ccievoicelab.c...@gmail.com Subject: [OSL | CCIE_Voice] Call Preservation Issue To: ccie_voice@onlinestudylist.com Date: Wednesday, June 1, 2011, 3:03 AM Hello Experts An interesting problem to solve for which I'm having a hard time figuring out the solution myself. We have a Call Manager node and IP Phones in one location and ingress voice gateways in other location. Both locations are connected over redundant WAN links. When one WAN link fails, it takes upto 5 seconds for the redundant WAN link to kick in. Issue is, if there is an active call between the gateway and the IP Phone, the call simply drops when a WAN issue occurs. Is there a way to preserve the RTP stream on the GW itself that can be resumed once the redundant WAN link takes over? Any input is appreciated! -Inline Attachment Follows- ___ 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] CUC Phone View
Sorry for late answer, but just thought about one more thing, make sure the phone has CTI Control enabled, without this if you even have phone association with the correct user it will not work. Same goes for all other services as voiceview on cue or ippm of cups. Regards, Sent from my iPad On 29 May 2011, at 20:32, Ki Wi kiwi.vo...@gmail.com wrote: Everything should be in place. I have manually matched the phone configuration as well. From HQ Ph1, i can login as others and still have option 4 available. It seems like this phone have some special configuration which others doesn't have. Anyway, it's ok. Gonna wipe out the configuration soon and try everything again. =) Thanks everyone! On Mon, May 30, 2011 at 2:52 AM, Vik Malhi vma...@ipexpert.com wrote: Just double check that you actually have a voicemail for the other users that are trying to use phoneview. Disable/enable globally and per user. Other than that- check the Authentication URL in the Enterprise params. Ensure hostname is replaced by UCM IPAddr. Seems to be a little flaky in this release. -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Wireless, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ From: Ki Wi kiwi.vo...@gmail.com Date: Sun, 29 May 2011 21:38:31 +0800 To: George Goglidze gogli...@gmail.com Cc: OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CUC Phone View I have associated every phones into that ID. Since lab is about speed, i don't think i want to waste time to pinpoint those mac address in use. On Sun, May 29, 2011 at 9:07 PM, George Goglidze gogli...@gmail.com wrote: HQ phone 1 must be associated to the application user that is configured for UC phone view. Check the others too, Sent from my iPad On 29 May 2011, at 11:34, Ki Wi kiwi.vo...@gmail.com wrote: anyone tried it? I followed the guide closely from here http://blog.ipexpert.com/2010/11/17/setting-up-phone-view/ My HQ Ph1 can choose option 5 then follow by 4 to view the messages while other phones such as HQ Ph2 and BR1 Ph1/2 didn't have the option 4 same as a lot of others in the comment. I used voicemail template so i deleted HQ Ph1 and HQ Ph2 and recreate them again. The result is the same, HQ Ph1 can do phoneview and view the messages while others can't. Any tips on it? ___ 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] Call Preservation Issue
Thanks Amit but unfortunately this will not work. As I mentioned before, the RTP stream is being interrupted too. H323 call preservation, in my opinion, is only for signaling loss to the Call Manager only, but here the IP Phones are temporarily losing connectivity to the GW too which will interrupt the RTP stream. On Tue, May 31, 2011 at 7:54 PM, amit batra batraji...@yahoo.com wrote: try this http://ccieash.wordpress.com/2010/06/25/h323-call-preservation/ might help you... --- On *Wed, 6/1/11, CCIE Voice ccievoicelab.c...@gmail.com* wrote: From: CCIE Voice ccievoicelab.c...@gmail.com Subject: Re: [OSL | CCIE_Voice] Call Preservation Issue To: amit batra batraji...@yahoo.com Cc: ccie_voice@onlinestudylist.com Date: Wednesday, June 1, 2011, 4:34 AM Sorry, should have been more clear. Using H.323. The call preservation feature of H323 applies when the gateway loses connectivity to Call Manager only, so that the RTP stream is preserved to the phone, but here the RTP stream itself is being interrupted. On May 31, 2011, at 6:58 PM, amit batra batraji...@yahoo.comhttp://mc/compose?to=batraji...@yahoo.com wrote: What type of protocol you are using ? H323 or MGCP ? With MGCP i dont think if there is any way .. H323 can preserve call .. --- On *Wed, 6/1/11, CCIE VOICE ccievoicelab.c...@gmail.comhttp://mc/compose?to=ccievoicelab.c...@gmail.com * wrote: From: CCIE VOICE ccievoicelab.c...@gmail.comhttp://mc/compose?to=ccievoicelab.c...@gmail.com Subject: [OSL | CCIE_Voice] Call Preservation Issue To: http://mc/compose?to=ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.comhttp://mc/compose?to=ccie_voice@onlinestudylist.com Date: Wednesday, June 1, 2011, 3:03 AM Hello Experts An interesting problem to solve for which I'm having a hard time figuring out the solution myself. We have a Call Manager node and IP Phones in one location and ingress voice gateways in other location. Both locations are connected over redundant WAN links. When one WAN link fails, it takes upto 5 seconds for the redundant WAN link to kick in. Issue is, if there is an active call between the gateway and the IP Phone, the call simply drops when a WAN issue occurs. Is there a way to preserve the RTP stream on the GW itself that can be resumed once the redundant WAN link takes over? Any input is appreciated! -Inline Attachment Follows- ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.comwww.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out http://www.PlatinumPlacement.comwww.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] Call Preservation Issue
try this http://ccieash.wordpress.com/2010/06/25/h323-call-preservation/ might help you... --- On Wed, 6/1/11, CCIE Voice ccievoicelab.c...@gmail.com wrote: From: CCIE Voice ccievoicelab.c...@gmail.com Subject: Re: [OSL | CCIE_Voice] Call Preservation Issue To: amit batra batraji...@yahoo.com Cc: ccie_voice@onlinestudylist.com Date: Wednesday, June 1, 2011, 4:34 AM Sorry, should have been more clear. Using H.323. The call preservation feature of H323 applies when the gateway loses connectivity to Call Manager only, so that the RTP stream is preserved to the phone, but here the RTP stream itself is being interrupted. On May 31, 2011, at 6:58 PM, amit batra batraji...@yahoo.com wrote: What type of protocol you are using ? H323 or MGCP ? With MGCP i dont think if there is any way .. H323 can preserve call .. --- On Wed, 6/1/11, CCIE VOICE ccievoicelab.c...@gmail.com wrote: From: CCIE VOICE ccievoicelab.c...@gmail.com Subject: [OSL | CCIE_Voice] Call Preservation Issue To: ccie_voice@onlinestudylist.com Date: Wednesday, June 1, 2011, 3:03 AM Hello Experts An interesting problem to solve for which I'm having a hard time figuring out the solution myself. We have a Call Manager node and IP Phones in one location and ingress voice gateways in other location. Both locations are connected over redundant WAN links. When one WAN link fails, it takes upto 5 seconds for the redundant WAN link to kick in. Issue is, if there is an active call between the gateway and the IP Phone, the call simply drops when a WAN issue occurs. Is there a way to preserve the RTP stream on the GW itself that can be resumed once the redundant WAN link takes over? Any input is appreciated! -Inline Attachment Follows- ___ 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] Problem With CUE
Hello everyone, I got two CUE from a friend, but one of then is rebooting the router. Hardware issue? there is a way to test it with parts of the one working? 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
Re: [OSL | CCIE_Voice] Call Preservation Issue
Voice service VoIP H323 Call preserve Should be the command set. I use it as my default configuration. Sent from my iPhone On May 31, 2011, at 4:04 PM, CCIE Voice ccievoicelab.c...@gmail.com wrote: Sorry, should have been more clear. Using H.323. The call preservation feature of H323 applies when the gateway loses connectivity to Call Manager only, so that the RTP stream is preserved to the phone, but here the RTP stream itself is being interrupted. On May 31, 2011, at 6:58 PM, amit batra batraji...@yahoo.com wrote: What type of protocol you are using ? H323 or MGCP ? With MGCP i dont think if there is any way .. H323 can preserve call .. --- On Wed, 6/1/11, CCIE VOICE ccievoicelab.c...@gmail.com wrote: From: CCIE VOICE ccievoicelab.c...@gmail.com Subject: [OSL | CCIE_Voice] Call Preservation Issue To: ccie_voice@onlinestudylist.com Date: Wednesday, June 1, 2011, 3:03 AM Hello Experts An interesting problem to solve for which I'm having a hard time figuring out the solution myself. We have a Call Manager node and IP Phones in one location and ingress voice gateways in other location. Both locations are connected over redundant WAN links. When one WAN link fails, it takes upto 5 seconds for the redundant WAN link to kick in. Issue is, if there is an active call between the gateway and the IP Phone, the call simply drops when a WAN issue occurs. Is there a way to preserve the RTP stream on the GW itself that can be resumed once the redundant WAN link takes over? Any input is appreciated! -Inline Attachment Follows- ___ 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] AIM-CUE CF card problem
Alex; You need to re-image your CF with another known good CF. I just did this several weeks ago for a UC500 system. I got another CF from a good system, then I used my linux server to do a bit by bit copy of the CF using 2 USB multi-card readers. If you are not familiar with linux you can use the Ultimate Boot CD (partitioning as well as other utilities). So the hard thing might be getting a known good CF. Sam. On Tue, May 31, 2011 at 5:43 PM, ccieid1ot ccieid...@gmail.com wrote: The mem is your ram, CF is your hd. Try the sandisk CF, I'm sure cisco just oem it from either sandisk or another manufacturer. duy ccie #27737 voice tmobile g2 On May 31, 2011 1:22 PM, Alex Goh ncsalex@gmail.com wrote: Hi Guys, Hope I can seek a little help here, my AIM-CUE 1GB CF card failed on me just 1 week before my exam! I've getting the error of Not a cisco supported CF. Please use cisco supported CF and reinstall the software. System Halted. Anyone know how to solved this issue? I've try to reinstall CUE using the boothelper, but no luck. Possibly the CF card is gone case. A search on google mentioned Cisco AIM-CUE check on the CF Card sector size, else refuse to work. But the used 1GB CF card was asking half the price of the AIM-CUE module /w 1GB CF itself on ebay :( It is anyway I can used on 3rd party CF card? saw it also certain SANDISK CF might work, but I'm not sure it is still able to find in the market now. Also, I notice the router Memory CF (MEM-CF-1GB) is selling cheaper than AIM-CUE-1GBCF, I wonder will it able to use? Any help will be appreciated. Regards, Alex ___ 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] T1 Interface question
Guys; I've been using a VWIC-2MFT-T1-DI in my PSTN/WAN router for the last 6 months. No issues encountered so far. Also for E1 cards, you can use a VWIC-(2)MFT-G.703. I'm using one in my BR2 Router. No special configs for either. For voice they act just the same as the regular cards. Hope that helps. Sam. On Mon, May 30, 2011 at 10:35 PM, Abel ... midga...@gmail.com wrote: There something we should know about the VWIC-2MFT-T1-DI, cause is a little cheaper than the normal one. Can we use the DI like the normal one? I was wondering about that some time ago. Thanks On Mon, May 30, 2011 at 8:49 PM, Bill Lake whl...@gmail.com wrote: VWIC-2MFT-T1 is a flex port that supports data and voice VWIC-2MFT-T1-DI adds add/drop multiplexing VWIC-2MFT-T1-DIR looks to add drop and insert and relays but might be a specialty product as I only see it listed for 1941-DC Mobile wireless edge router On Mon, May 30, 2011 at 6:42 PM, Bill Hatcher wchatc...@gmail.comwrote: I am trying to build a CCIE Voice lab at my house. I am looking at equipment on eBay and nned to know what is the difference between VWIC-2MFT-T1-DI and VWIC-2MFT-T1-DIR and can I use these inplace of a VWIC-2MFT-T1? ___ 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
[OSL | CCIE_Voice] LAN QoS
hello all, I have a question for you guys hopefully you can help me understand, this is way too confusing for me when i issue auto qos on a 3750 switch this is how the config will look like mls qos srr-queue output cos-map queue 1 threshold 3 5 mls qos srr-queue output cos-map queue 2 threshold 3 3 6 7 mls qos srr-queue output cos-map queue 3 threshold 3 2 4 mls qos srr-queue output cos-map queue 4 threshold 2 1 mls qos srr-queue output cos-map queue 4 threshold 3 0 mls qos srr-queue output dscp-map queue 1 threshold 3 40 41 42 43 44 45 46 47 mls qos srr-queue output dscp-map queue 2 threshold 3 24 25 26 27 28 29 30 31 mls qos srr-queue output dscp-map queue 2 threshold 3 48 49 50 51 52 53 54 55 mls qos srr-queue output dscp-map queue 2 threshold 3 56 57 58 59 60 61 62 63 mls qos srr-queue output dscp-map queue 3 threshold 3 16 17 18 19 20 21 22 23 mls qos srr-queue output dscp-map queue 3 threshold 3 32 33 34 35 36 37 38 39 mls qos srr-queue output dscp-map queue 4 threshold 1 8 mls qos srr-queue output dscp-map queue 4 threshold 2 9 10 11 12 13 14 15 mls qos srr-queue output dscp-map queue 4 threshold 3 0 1 2 3 4 5 6 7 mls qos queue-set output 1 threshold 1 138 138 92 138 mls qos queue-set output 1 threshold 2 138 138 92 400 mls qos queue-set output 1 threshold 3 36 77 100 318 mls qos queue-set output 1 threshold 4 20 50 67 400 mls qos queue-set output 2 threshold 1 149 149 100 149 mls qos queue-set output 2 threshold 2 118 118 100 235 mls qos queue-set output 2 threshold 3 41 68 100 272 mls qos queue-set output 2 threshold 4 42 72 100 242 mls qos queue-set output 1 buffers 10 10 26 54 mls qos queue-set output 2 buffers 16 6 17 61 mls qos when i'm asked to change the COS or DSCP values from Q to Q how do I know which threshold to use ? for exambple if i'm asked to configure the COS valus 2,3 to Queue 4 how do i know on which threshold withing Q4 ? does my question make sense ? this part of the threshold is very confusing for me hopefully you can help me out 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
Re: [OSL | CCIE_Voice] IPPM on cisco7961 didn't alert
Ya, i believe assigned the standard CTI control of all devices + standard CTI enabled to phone messenger apps user will do the trick. For phone user,a combination of standard CCM end user + standard cti enabled should be sufficient. Shingei. On Wed, Jun 1, 2011 at 6:11 AM, Ki Wi kiwi.vo...@gmail.com wrote: Finally found the problem, I didn't give enough right to IPPM user. Seems like it need CCM Super user right instead! On Wed, Jun 1, 2011 at 6:07 AM, Ki Wi kiwi.vo...@gmail.com wrote: For me, everything is fine. I just do everything again from fresh. Same problem. Something must be missing along the way. Is there any IPPM guide online? The one i found from http://docwiki.cisco.com/wiki/Cisco_Unified_Presence%2C_Release_7.x_--_Configuring_Cisco_IP_Phone_Messenger_on_Cisco_Unified_Presenceis pretty useless. On Wed, Jun 1, 2011 at 1:21 AM, Alex Goh ncsalex@gmail.com wrote: Hi Ki Wi, I've encounter the same issue also, and I solved it by changing the Enterprise Parameters Services URL to IP instead of hostname (Apparently, I miss that part when I reverted my VMware snapshot), remember I saw this solution from OSL discussion before. HTH Cheers, Alex On Tue, May 31, 2011 at 11:25 AM, ShinGei Yong shingei.y...@gmail.comwrote: Frens, If i can recall correctly, that was due to that i missed associate the phone with application user Phone Messenger. You need the phone messenger application user to control the IPPM user. Without the association, the messaging will still work but funny stuff come out, if not wrong Shingei. On Tue, May 31, 2011 at 4:52 AM, Ki Wi kiwi.vo...@gmail.com wrote: Hey, Do you still remember how did you resolve this alert issue? I'm still trying to train myself up in CUPS. Last night, my alert was working, my IPPM login wasn't. Today my IPPM is working but no alert. =( All other components are working. On Sun, Dec 26, 2010 at 12:59 AM, ShinGei Yong shingei.y...@gmail.com wrote: Guys, Pls ignore this mail, has managed to figured out the caused. thanks Shingei. On Sat, Dec 25, 2010 at 4:36 PM, ShinGei Yong shingei.y...@gmail.com wrote: Hi, I've configure the IPPM on cisco 7961 phone, everything works smooth other that the message receive alert. It doesn't ring when there is a mgs come in from CIPC or other IPPM.i've set the audible alert to ON but still got no luck. Another IPPM phone encounter the same issue, so don't think is the phone problem. Any idea? Thanks Shingei. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com 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] T1 Interface question
Great proof of concept, they should work exactly the same since they only add features but you never know until you do it : ) On Tue, May 31, 2011 at 7:51 PM, Sam Park upperlevelpark...@gmail.comwrote: Guys; I've been using a VWIC-2MFT-T1-DI in my PSTN/WAN router for the last 6 months. No issues encountered so far. Also for E1 cards, you can use a VWIC-(2)MFT-G.703. I'm using one in my BR2 Router. No special configs for either. For voice they act just the same as the regular cards. Hope that helps. Sam. On Mon, May 30, 2011 at 10:35 PM, Abel ... midga...@gmail.com wrote: There something we should know about the VWIC-2MFT-T1-DI, cause is a little cheaper than the normal one. Can we use the DI like the normal one? I was wondering about that some time ago. Thanks On Mon, May 30, 2011 at 8:49 PM, Bill Lake whl...@gmail.com wrote: VWIC-2MFT-T1 is a flex port that supports data and voice VWIC-2MFT-T1-DI adds add/drop multiplexing VWIC-2MFT-T1-DIR looks to add drop and insert and relays but might be a specialty product as I only see it listed for 1941-DC Mobile wireless edge router On Mon, May 30, 2011 at 6:42 PM, Bill Hatcher wchatc...@gmail.comwrote: I am trying to build a CCIE Voice lab at my house. I am looking at equipment on eBay and nned to know what is the difference between VWIC-2MFT-T1-DI and VWIC-2MFT-T1-DIR and can I use these inplace of a VWIC-2MFT-T1? ___ 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] IPPM on cisco7961 didn't alert
Maybe you can try it next time, for me it doesn't alert me till I give me phonemessenger(IPPM) account a CCM super user right, weird. On Wed, Jun 1, 2011 at 9:26 AM, ShinGei Yong shingei.y...@gmail.com wrote: Ya, i believe assigned the standard CTI control of all devices + standard CTI enabled to phone messenger apps user will do the trick. For phone user,a combination of standard CCM end user + standard cti enabled should be sufficient. Shingei. On Wed, Jun 1, 2011 at 6:11 AM, Ki Wi kiwi.vo...@gmail.com wrote: Finally found the problem, I didn't give enough right to IPPM user. Seems like it need CCM Super user right instead! On Wed, Jun 1, 2011 at 6:07 AM, Ki Wi kiwi.vo...@gmail.com wrote: For me, everything is fine. I just do everything again from fresh. Same problem. Something must be missing along the way. Is there any IPPM guide online? The one i found from http://docwiki.cisco.com/wiki/Cisco_Unified_Presence%2C_Release_7.x_--_Configuring_Cisco_IP_Phone_Messenger_on_Cisco_Unified_Presenceis pretty useless. On Wed, Jun 1, 2011 at 1:21 AM, Alex Goh ncsalex@gmail.com wrote: Hi Ki Wi, I've encounter the same issue also, and I solved it by changing the Enterprise Parameters Services URL to IP instead of hostname (Apparently, I miss that part when I reverted my VMware snapshot), remember I saw this solution from OSL discussion before. HTH Cheers, Alex On Tue, May 31, 2011 at 11:25 AM, ShinGei Yong shingei.y...@gmail.comwrote: Frens, If i can recall correctly, that was due to that i missed associate the phone with application user Phone Messenger. You need the phone messenger application user to control the IPPM user. Without the association, the messaging will still work but funny stuff come out, if not wrong Shingei. On Tue, May 31, 2011 at 4:52 AM, Ki Wi kiwi.vo...@gmail.com wrote: Hey, Do you still remember how did you resolve this alert issue? I'm still trying to train myself up in CUPS. Last night, my alert was working, my IPPM login wasn't. Today my IPPM is working but no alert. =( All other components are working. On Sun, Dec 26, 2010 at 12:59 AM, ShinGei Yong shingei.y...@gmail.com wrote: Guys, Pls ignore this mail, has managed to figured out the caused. thanks Shingei. On Sat, Dec 25, 2010 at 4:36 PM, ShinGei Yong shingei.y...@gmail.com wrote: Hi, I've configure the IPPM on cisco 7961 phone, everything works smooth other that the message receive alert. It doesn't ring when there is a mgs come in from CIPC or other IPPM.i've set the audible alert to ON but still got no luck. Another IPPM phone encounter the same issue, so don't think is the phone problem. Any idea? Thanks Shingei. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com 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
Use the auto configured threshold is fine and modify the cos of the particular q. What you are looking at is the dscp-map to q. duy ccie #27737 voice tmobile g2 On May 31, 2011 9:21 PM, Cristobal Priego cristobalpri...@gmail.com wrote: hello all, I have a question for you guys hopefully you can help me understand, this is way too confusing for me when i issue auto qos on a 3750 switch this is how the config will look like mls qos srr-queue output cos-map queue 1 threshold 3 5 mls qos srr-queue output cos-map queue 2 threshold 3 3 6 7 mls qos srr-queue output cos-map queue 3 threshold 3 2 4 mls qos srr-queue output cos-map queue 4 threshold 2 1 mls qos srr-queue output cos-map queue 4 threshold 3 0 mls qos srr-queue output dscp-map queue 1 threshold 3 40 41 42 43 44 45 46 47 mls qos srr-queue output dscp-map queue 2 threshold 3 24 25 26 27 28 29 30 31 mls qos srr-queue output dscp-map queue 2 threshold 3 48 49 50 51 52 53 54 55 mls qos srr-queue output dscp-map queue 2 threshold 3 56 57 58 59 60 61 62 63 mls qos srr-queue output dscp-map queue 3 threshold 3 16 17 18 19 20 21 22 23 mls qos srr-queue output dscp-map queue 3 threshold 3 32 33 34 35 36 37 38 39 mls qos srr-queue output dscp-map queue 4 threshold 1 8 mls qos srr-queue output dscp-map queue 4 threshold 2 9 10 11 12 13 14 15 mls qos srr-queue output dscp-map queue 4 threshold 3 0 1 2 3 4 5 6 7 mls qos queue-set output 1 threshold 1 138 138 92 138 mls qos queue-set output 1 threshold 2 138 138 92 400 mls qos queue-set output 1 threshold 3 36 77 100 318 mls qos queue-set output 1 threshold 4 20 50 67 400 mls qos queue-set output 2 threshold 1 149 149 100 149 mls qos queue-set output 2 threshold 2 118 118 100 235 mls qos queue-set output 2 threshold 3 41 68 100 272 mls qos queue-set output 2 threshold 4 42 72 100 242 mls qos queue-set output 1 buffers 10 10 26 54 mls qos queue-set output 2 buffers 16 6 17 61 mls qos when i'm asked to change the COS or DSCP values from Q to Q how do I know which threshold to use ? for exambple if i'm asked to configure the COS valus 2,3 to Queue 4 how do i know on which threshold withing Q4 ? does my question make sense ? this part of the threshold is very confusing for me hopefully you can help me out 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
