[OSL | CCIE_Voice] OFF TOPIC: Avaya ACS integration.
To All, I apologize for distracting you from your studying. I'm looking for some input from anyone with Avaya ACS experience. I'm trying to provide CME to Avaya Partner ACS integration while we migrate to CME. I currently have VIC-4FXO and VIC2-FXO connected to extension ports on the avaya. I'm able to call to the PSTN without issue, but can not dial internal two digit extensions. As I understand it , the user must press the intercom button prior to dialing an internal extension. A quick look at the avaya documentation indicates the system supports single-line pots phones and should be able to dial internal extensions by using a switch hook signal before dialing. I'm trying to figure out if this switch hook support needs to be enabled and also if enabling the flash button in CME would be the same as a switch hook to the avaya. Any ideas or opinions are greatly appreciated. ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
[OSL | CCIE_Voice] CUE voiceview authentication headache :)
Hi group, I have my CUE configured with CUCM. I can call into CUE voicemail and check messages, so I know my PIN is good. I've subscribed my phone to the voiceview service and can get the authentication url to show on the phone. I login in with username scphX (thats the username) and the pin, but I get login failed everytime. Can anyone help me understand where the voiceview app is referencing for authentication? Does it change if the CUE module it integrated to CME or CUCM ? Thanks 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
[OSL | CCIE_Voice] Question regarding dialing NATIONAL calls outside the U.S.
Hi All, It's probably a silly question, but it's some I struggle with more often than I like. Can some clarify for me how a LOCAL and NATIONAL call generally need to be dialed / presented outside the U.S. In all the practice labs I've done, dialing international is usually straight forward - something like 90014085551212. But my head spins when trying to understand what is a national call verus a local call. Lets say I have a International e164 phone number at +743 2202 4xxx at my Site C location. Is a local number anything in the format? What would a National number in this example ? It seems to me anything other than , would be a International call. So I don't see where National fits in. Also , when I see a number like XX on the PSTN phone, is that automatically a International call ? - 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
[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
Re: [OSL | CCIE_Voice] MVA -Your call can not be completed as dialed
Thank you Vipul. I thought re-routing CSS was used only for the SNR configuration. I will take a look and test again. :) On Tue, Aug 7, 2012 at 3:29 PM, Vipul Jindal (vipjinda) vipji...@cisco.comwrote: 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] MVA -Your call can not be completed as dialed
Actually, the MVA Prompt was working. I believe the issue was a owner/device mismatch actually. I did try to adjust the re-route CSS but it didn't change the behavior. I actually deleted the user (which surprising wipes all the remote destination info for the user also) and recreated it. It's all working as expected now. Thanks for the clarification regarding Re-routing CSS. I didn't think it would apply here. On Tue, Aug 7, 2012 at 4:18 PM, Krishna vinayak_...@yahoo.com wrote: 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 ___ 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
Ramy, Can you clarify how the re-routing CSS is used? Is the regular CSS ignored when a remote destination phone is dialing through MVA? -Scott On Tue, Aug 7, 2012 at 4:34 PM, Ramy Abdelrahim ramyoth...@hotmail.comwrote: Guys, Re-routing CSS is used for SNR but CSS is used for MVA. Therefore, you've to check if the number you're dialing is in a route pattern that the configured CSS in the remote destination profile can reach. I recommend to separate the dial plan for MVA or SNR as follows. - Create a new partition. - Create a new CSS containing the above new partition. - Create a new route list - Create a new route pattern. hope this will help. Regards, Ramy -- Date: Tue, 7 Aug 2012 16:21:17 -0400 From: ccielab...@gmail.com To: vipji...@cisco.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MVA -Your call can not be completed as dialed b Thank you Vipul. I thought re-routing CSS was used only for the SNR configuration. I will take a look and test again. :) On Tue, Aug 7, 2012 at 3:29 PM, Vipul Jindal (vipjinda) vipji...@cisco.com wrote: 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 ___ 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] CUE and SIP-Notify
Just working on some CUE stuff today and stumbled across the option to configure dtmf-relay rtp-nte under the sip subsystem of the CUE. In general , I've always followed the consensus that you shoud use RTP-NTE everywhere BUT when communicating with CUE which uses SIP-NOTIFY. I wanted to reach out the group to see if anyone has input on any of the implications of changing the SIP subsystem to use RTP-NTE instead of SIP-NOTIFY ___ 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] BACD using builtin: services.
I'm trying to use the newer builtin b-acd services in my lab setup. Can anyone confirm my config and see if there is anything I'm missing? I'm using the usual B-acd documentation as a template. I'm getting the a problem where the call drops as soon as I call the pilot number application service app-b-acd-aa paramspace english index 0 param max-time-call-retry 700 param voice-mail 4001 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 app-b-acd-aa-pilot 4000 param drop-through-prompt _dt_prompt.au param second-greeting-time 60 param call-retry-timer 15 ! service app-b-acd param queue-len 10 param number-of-hunt-grps 1 param queue-manager-debugs 1 ! dial-peer voice 4000 pots service app-b-acd-aa incoming called-number 4000 direct-inward-dial ___ 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] BACD using builtin: services.
Nevermind I messed so much on the previous attempt , it's not worth looking at. :) Here is a config that seems to work. application service app-b-acd param queue-len 10 param aa-hunt1 4010 param number-of-hunt-grps 1 param queue-manager-debugs 1 ! service app-b-acd-aa paramspace english index 0 param number-of-hunt-grps 1 param drop-through-option 1 param handoff-string app-b-acd-aa paramspace english language en param max-time-vm-retry 2 param aa-pilot 4000 paramspace english location flash:/ param second-greeting-time 60 param drop-through-prompt _dt_prompt.au param call-retry-timer 15 param voice-mail 4001 param max-time-call-retry 700 param service-name app-b-acd ! dial-peer voice 4000 pots service app-b-acd-aa incoming called-number 4000 direct-inward-dial On Sat, Dec 10, 2011 at 3:11 PM, ccielabrat ccielab...@gmail.com wrote: I'm trying to use the newer builtin b-acd services in my lab setup. Can anyone confirm my config and see if there is anything I'm missing? I'm using the usual B-acd documentation as a template. I'm getting the a problem where the call drops as soon as I call the pilot number application service app-b-acd-aa paramspace english index 0 param max-time-call-retry 700 param voice-mail 4001 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 app-b-acd-aa-pilot 4000 param drop-through-prompt _dt_prompt.au param second-greeting-time 60 param call-retry-timer 15 ! service app-b-acd param queue-len 10 param number-of-hunt-grps 1 param queue-manager-debugs 1 ! dial-peer voice 4000 pots service app-b-acd-aa incoming called-number 4000 direct-inward-dial ___ 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] UCCX and transcoding
So if that is the case, I guess it makes sense that the RP would need to be in the HQ device pool , to force the call to g.729 based on the GW being in the SiteB region and the RP being in the HQ region. I will test that today. Strange thing is if I call the RP from a phone in SiteB , the codec gets set properly. It's only when I call from PSTN via SiteB's Gw that the codec doesn't get forced to g.729 as I would expect. On Fri, Dec 9, 2011 at 3:01 AM, datucha123 datucha123 datucha...@gmail.comwrote: Well I think you need to use the transcoding based on the CTI ports, not the Route Point, as the RP just forwards calls, there are not RTP stream ever to Route Point. On Fri, Dec 9, 2011 at 8:23 AM, ccielabrat ccielab...@gmail.com wrote: I need some help clarifying where I need transcoding for a UCCX where the trigger DN is located across the WAN (Site B) I have the MGCP GW in Site B in a Device Pool (SiteB) UCCX CTI-RP is in Device Pool (SiteB) and CTI-Ports are all in Device Pool (HQ) Per my region configuration, g.729 is required between HQ and SiteB. My expectation is the call will arrive in SiteB and based on region requirements, will be setup as a g.729 call. A transcoder would be needed at the UCCX side to get to g.711. Would I be correct in thinking the CTI-Port should invoke trancoding out of the MRGL assigned to the HQ dev pool ? Currently the call comes in and I see the MGCP gateway with a g.711 call, so it appears it's going across the WAN as g.711. I know I'm missing some understanding here. 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 http://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] UCCX and transcoding
As I spend more time thinking about this, I think I'm complicating the whole scenario. As you mention, the RP is simply forwarding the call to a CTI Port. The only time a transcoder would have to be invoked is for the aef script when playing prompts. Otherwise the handoff to an agent would follow the normal regions/codec rules. On Fri, Dec 9, 2011 at 1:55 PM, datucha123 datucha123 datucha...@gmail.comwrote: I cannot understand why do you have CTI RP and CTI ports in a different Device Pool? Also please ensure that the Transcoder is assigned the correct Device Pool - HQ Device Pool. Let me test it in my LAB tommorow as well, and will infrom you. On Fri, Dec 9, 2011 at 4:50 PM, ccielabrat ccielab...@gmail.com wrote: So if that is the case, I guess it makes sense that the RP would need to be in the HQ device pool , to force the call to g.729 based on the GW being in the SiteB region and the RP being in the HQ region. I will test that today. Strange thing is if I call the RP from a phone in SiteB , the codec gets set properly. It's only when I call from PSTN via SiteB's Gw that the codec doesn't get forced to g.729 as I would expect. On Fri, Dec 9, 2011 at 3:01 AM, datucha123 datucha123 datucha...@gmail.com wrote: Well I think you need to use the transcoding based on the CTI ports, not the Route Point, as the RP just forwards calls, there are not RTP stream ever to Route Point. On Fri, Dec 9, 2011 at 8:23 AM, ccielabrat ccielab...@gmail.com wrote: I need some help clarifying where I need transcoding for a UCCX where the trigger DN is located across the WAN (Site B) I have the MGCP GW in Site B in a Device Pool (SiteB) UCCX CTI-RP is in Device Pool (SiteB) and CTI-Ports are all in Device Pool (HQ) Per my region configuration, g.729 is required between HQ and SiteB. My expectation is the call will arrive in SiteB and based on region requirements, will be setup as a g.729 call. A transcoder would be needed at the UCCX side to get to g.711. Would I be correct in thinking the CTI-Port should invoke trancoding out of the MRGL assigned to the HQ dev pool ? Currently the call comes in and I see the MGCP gateway with a g.711 call, so it appears it's going across the WAN as g.711. I know I'm missing some understanding here. 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 http://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] Calling party transformations on a gateway
Thanks Chris, I'll look into that now. So , with this in mind, should I be able to check the use external # mask within the calling party transformation and then apply a XX mask? or do I have to match against the 4 digit DN and then prefix the DN in the mask? On Thu, Dec 8, 2011 at 8:42 AM, Chris Martin clm.c...@gmail.com wrote: What pattern do you have setup for Calling Party Transformations? You mention you are seeing + Globalized numbers in your isdn q931 debug, this leads me to believe you have a RP/RL with External Number Mask applied. My guess is you are trying to match on the Globalized Calling numbers in your calling transformation patterns. The calling party transformation patterns need to be matched on the pre-RP/RL digits. IE: 5XXX not +12127775XXX. HTH, Chris On Wed, Dec 7, 2011 at 10:13 PM, ccielabrat ccielab...@gmail.com wrote: To All, I just needed to check to see if anyone knows about a problem using calling party transformations at the gateway level. I have a setup where I am send a fully globalized called and calling # to my gateways. I wanted to make all needed adjustments just before it goes to the PSTN. My transformation CSS's are setup properly and I can adjust the called party number but the calling party number will not adjust according to the calling party xform pattern. I can see the calling number is getting to the gw in + format based on what I see in Q.931 on the router. Anybody experience this behavior? ___ 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] Calling party transformations on a gateway
Hey Chris, Great catch. You were spot on with your advice. Thanks for correcting my thinking. I guess it doesn't make sense as I think about it now, but I thought by the time the gateway got the call had left CUCM and any reference to external # mask, etc. wouldn't be relevant. So I was trying to get the send the + globalized # to the gw and have it trim off the +. On Thu, Dec 8, 2011 at 8:50 AM, ccielabrat ccielab...@gmail.com wrote: Thanks Chris, I'll look into that now. So , with this in mind, should I be able to check the use external # mask within the calling party transformation and then apply a XX mask? or do I have to match against the 4 digit DN and then prefix the DN in the mask? On Thu, Dec 8, 2011 at 8:42 AM, Chris Martin clm.c...@gmail.com wrote: What pattern do you have setup for Calling Party Transformations? You mention you are seeing + Globalized numbers in your isdn q931 debug, this leads me to believe you have a RP/RL with External Number Mask applied. My guess is you are trying to match on the Globalized Calling numbers in your calling transformation patterns. The calling party transformation patterns need to be matched on the pre-RP/RL digits. IE: 5XXX not +12127775XXX. HTH, Chris On Wed, Dec 7, 2011 at 10:13 PM, ccielabrat ccielab...@gmail.com wrote: To All, I just needed to check to see if anyone knows about a problem using calling party transformations at the gateway level. I have a setup where I am send a fully globalized called and calling # to my gateways. I wanted to make all needed adjustments just before it goes to the PSTN. My transformation CSS's are setup properly and I can adjust the called party number but the calling party number will not adjust according to the calling party xform pattern. I can see the calling number is getting to the gw in + format based on what I see in Q.931 on the router. Anybody experience this behavior? ___ 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] UCCX on VMWARE problems.
I just wanted to do some testing on UCCX so I booted a vmware image of UCCX that I've used before. It's a fresh install with no integration. When I log in, it says the JTAPI is out of sync. I've fixed the JTAPI problem related to moving C:\windows\java files to c:\winnt\java It wants me to rerun jtapi sync, but the menus on the UCCX page page will not display correctly. Do dropdown menus appear. Can anyone help? ___ 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] UCCX on VMWARE problems.
I can't get to the point to upload the license. On Thu, Dec 8, 2011 at 2:07 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote: do you have the License uploaded to the CCX , this can happen when you have no license Ash On Thu, Dec 8, 2011 at 12:20 PM, ccielabrat ccielab...@gmail.com wrote: I just wanted to do some testing on UCCX so I booted a vmware image of UCCX that I've used before. It's a fresh install with no integration. When I log in, it says the JTAPI is out of sync. I've fixed the JTAPI problem related to moving C:\windows\java files to c:\winnt\java It wants me to rerun jtapi sync, but the menus on the UCCX page page will not display correctly. Do dropdown menus appear. Can anyone help? ___ 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] UCCX on VMWARE problems.
Hi Ash, I stand corrected, I had uploaded the license file. I was at the point right after the initial setup completes and prompts you to close your browser. At that point, it presented the jtapi error. I got it working though. I ran setup /x from the UCCX media image and uninstalled UCCX. I then re-ran setup and have a clean UCCX server to mess with. Thank you as always for jumping in to help. On Thu, Dec 8, 2011 at 5:02 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote: What do you mean ? Where are you at exactly in the CCX installation ? Screenshot? Ash On Thursday, December 8, 2011, ccielabrat ccielab...@gmail.com wrote: I can't get to the point to upload the license. On Thu, Dec 8, 2011 at 2:07 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote: do you have the License uploaded to the CCX , this can happen when you have no license Ash On Thu, Dec 8, 2011 at 12:20 PM, ccielabrat ccielab...@gmail.com wrote: I just wanted to do some testing on UCCX so I booted a vmware image of UCCX that I've used before. It's a fresh install with no integration. When I log in, it says the JTAPI is out of sync. I've fixed the JTAPI problem related to moving C:\windows\java files to c:\winnt\java It wants me to rerun jtapi sync, but the menus on the UCCX page page will not display correctly. Do dropdown menus appear. Can anyone help? ___ 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] UCCX and transcoding
I need some help clarifying where I need transcoding for a UCCX where the trigger DN is located across the WAN (Site B) I have the MGCP GW in Site B in a Device Pool (SiteB) UCCX CTI-RP is in Device Pool (SiteB) and CTI-Ports are all in Device Pool (HQ) Per my region configuration, g.729 is required between HQ and SiteB. My expectation is the call will arrive in SiteB and based on region requirements, will be setup as a g.729 call. A transcoder would be needed at the UCCX side to get to g.711. Would I be correct in thinking the CTI-Port should invoke trancoding out of the MRGL assigned to the HQ dev pool ? Currently the call comes in and I see the MGCP gateway with a g.711 call, so it appears it's going across the WAN as g.711. I know I'm missing some understanding here. 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
[OSL | CCIE_Voice] Calling party transformations on a gateway
To All, I just needed to check to see if anyone knows about a problem using calling party transformations at the gateway level. I have a setup where I am send a fully globalized called and calling # to my gateways. I wanted to make all needed adjustments just before it goes to the PSTN. My transformation CSS's are setup properly and I can adjust the called party number but the calling party number will not adjust according to the calling party xform pattern. I can see the calling number is getting to the gw in + format based on what I see in Q.931 on the router. Anybody experience this behavior? ___ 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] Dtmf problem with MVA
I have MVA configured on my h323 router, with the appropriate dial peers as per the Cucm help pages. I am able to call into the piloting dn and I can hear the MVA application prompt me for my pin. When I press any digits, they are not recognized. I've tried to force a g711 codec and ensured Dtmf-relay is configured, but it doesn't change the problem. Has anyone run into this issue? ___ 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 question
I ran into a problem the other day that has me confused. I ran auto qos on the hq side, changed values as needed and pasted the modified policy in the hq router. I then took the same policy and pasted it into the SB router config and bound it to the dlci. All seemed to be ok until I tried to get phones registered. I could get a dhcp address but never register. I knew something with the WAN qos was screwed up. I've done it to myself in practice and in the real lab. It turns out that I didn't have frts on the physical interface. Once I put it on, everything started working. My questions are: 1.) I thought IOS would refuse attaching a class to a dlci without frts on the physical interface. 2.) Without frts configured on the physical interface, wouldn't the class assignment on the dlci (384k) effectively be ignored? I know what the config problem was, but I understand what it was actually causing to happen at the Pvc level. ___ 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 Strategy Question.
Hey everyone, I can confirm after A LOT of testing, if you are given a QOS requirement for only one of the two Frame PVCs, and you use Auto QOS, you will have a problem. Auto QOS will automatically config Frame-relay Traffic shaping on the physical WAN interface and then configure the PVC you are QOS'ing to the bandwidth that is noted under the sub interface. The other sub interface gets left with the default frame-relay traffic shaping behavior which is to drop CIR to 56k on the PVC. Do a Show Frame PVC dlci# on both PVC's after running auto qos. I think this could be an intended Rat hole on the exam. If you only have to configure Hq-SB QOS and you don't know much more the to run autoqos and tweak a couple parameters, your SC communications will start to fail with a PVC CIR of 56k. On Sat, Dec 3, 2011 at 10:00 AM, Ken Wyan kew...@gmail.com wrote: Hi, I think you got 56k value from this document which was published in 2005 with IOS version 11. (somehow same age as QoS SRND) http://www.cisco.com/en/US/tech/tk713/tk237/technologies_configuration_example09186a00800942f8.shtml I think it's better to put auto qos voip or auto qos voip fr-atm in the remaining interface as well (without any bandwidth as it's not mentioned in exam). Then itll take 1.5M by default. Is there a command to verify that FRTS use 56k bandwidth because above documents are very old. Ken On Sat, Dec 3, 2011 at 8:54 AM, ccielabrat ccielab...@gmail.com wrote: To All, I've been trying to figure the best/fastest way to get a WAN QOS requirement completed on exam day. I've become very comfortable with Auto-QOS and making the needed tweaks, so Auto-QOS is the way I'm going to use. The one piece of the strategy that I'm stilll wondering about is if WAN QOS is specified for only one of the PVC's. Auto-QOS will automatically put Frame-relay traffic shaping on the physical interface which has the side effect of leaving the other pvc with a 56k PVC speed. My solution here is to create a frame-relay map-class with the following parameters. map-class frame-relay Not56k frame-relay traffice-rate 1536 I apply this map-class to the other sub-interface/PVC which negates the 56k problem. I'm curious if anyone has an opinion on the Pros/Cons of this approach and if it might negate requirements somehow. ___ 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 http://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 Strategy Question.
Thanks for clarifying this Vik. I wasn't aware of the crtp one way audio trap. But I'll keep it in mind. One question : I thought we could avoid having the frame relay traffic shaping on the physical interface if we did generic traffic shaping via class-map within the frame-relay map-class. Is this not the case? I vaguely remember what you are talking about where I wasn't able to apply a class to a DLCI and it prompted a message asking for traffic shaping to be enabled. Just wondering if there is work around or is it a hard and fast rule that Frame traffic shaping MUST be on the physical interface. On Sat, Dec 3, 2011 at 4:15 PM, Vik Malhi vma...@ipexpert.com wrote: This is correct. It's not AutoQoS that causes the problem- its because you MUST have FRTS enabled in order for the map-class to be attached to the DLCI. And this must happen since the service policy is inside the map-class. I recommend you run AutoQoS on all routers when doing WAN QoS or at the very least attach a map-class to all DLCI's. Also be careful if you are using cRTP. You should ensure that if you are using cRTP that both ends of the pipe are configured with cRTP otherwise you will experience one way audio. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Dec 3, 2011, at 10:59 AM, ccielabrat wrote: Hey everyone, I can confirm after A LOT of testing, if you are given a QOS requirement for only one of the two Frame PVCs, and you use Auto QOS, you will have a problem. Auto QOS will automatically config Frame-relay Traffic shaping on the physical WAN interface and then configure the PVC you are QOS'ing to the bandwidth that is noted under the sub interface. The other sub interface gets left with the default frame-relay traffic shaping behavior which is to drop CIR to 56k on the PVC. Do a Show Frame PVC dlci# on both PVC's after running auto qos. I think this could be an intended Rat hole on the exam. If you only have to configure Hq-SB QOS and you don't know much more the to run autoqos and tweak a couple parameters, your SC communications will start to fail with a PVC CIR of 56k. On Sat, Dec 3, 2011 at 10:00 AM, Ken Wyan kew...@gmail.com wrote: Hi, I think you got 56k value from this document which was published in 2005 with IOS version 11. (somehow same age as QoS SRND) http://www.cisco.com/en/US/tech/tk713/tk237/technologies_configuration_example09186a00800942f8.shtml I think it's better to put auto qos voip or auto qos voip fr-atm in the remaining interface as well (without any bandwidth as it's not mentioned in exam). Then itll take 1.5M by default. Is there a command to verify that FRTS use 56k bandwidth because above documents are very old. Ken On Sat, Dec 3, 2011 at 8:54 AM, ccielabrat ccielab...@gmail.com wrote: To All, I've been trying to figure the best/fastest way to get a WAN QOS requirement completed on exam day. I've become very comfortable with Auto-QOS and making the needed tweaks, so Auto-QOS is the way I'm going to use. The one piece of the strategy that I'm stilll wondering about is if WAN QOS is specified for only one of the PVC's. Auto-QOS will automatically put Frame-relay traffic shaping on the physical interface which has the side effect of leaving the other pvc with a 56k PVC speed. My solution here is to create a frame-relay map-class with the following parameters. map-class frame-relay Not56k frame-relay traffice-rate 1536 I apply this map-class to the other sub-interface/PVC which negates the 56k problem. I'm curious if anyone has an opinion on the Pros/Cons of this approach and if it might negate requirements somehow. ___ 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 http://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] Need some help with GK/CUBE Remote Zone call debugging.
Hi Ash,
Thanks very much for taking the time to reply.
I would really like to understand all the pieces to this scenario.
The debug I posted was from the HQ router/GK/CUBE, I'm not sure how to read
it yet, so I can't say what call leg it represents.
We are talking about the same scenario you mention in your reply.
CUCM via GK-0Trunk (No MTP configured) (Wait for H245 unchecked)
GK configured with a Remote Zone using a outvia to local CUBE.
CUBE is configured with One inbound dial-peer 011! with a fixed codec of
g.711 and an outbound dial-peer targeting RAS (default g.729)
The call setup works and I can answer the call but the rtp never works.
Please confirm my understanding of the problem.
1.) CUCM does ARQ/setup via GK
2.) GK sends LRQ to BBGK and gets LCF that it's a routable DN
3.) GK tells CUCM to target CUBE for H.323 call setup because of Outvia
config for BBGK Zone.
4.) CUCM sends H.225 to CUBE which triggers CUBE to do H.225 to endpoint in
BBGK Zone.
5.) CUBE waits for h.245 TCS and doesn't send H.225 connect back to CUCM
6.) BBGK Endpoint doesn't send any TCS and causes CUBE to wait /timeout for
H.245
7.) Call fails with CUBE disconnecting both BBGK call leg and CUCM call leg.
I think this is due to the fact that CUCM (without MTP) is forced to do
slow start , while CUBE will automatically do fast start.
As I understand , CUBE can't compensate for the difference between
slow/fast start call legs.
So is the only option to have an MTP configured at CUCM side?
Can the CUBE be forced to do slow start? Would that fix the issue?
On Fri, Dec 2, 2011 at 1:20 AM, Ashraf Ayyash ash.ayy...@gmail.com wrote:
What are you looking at in the debugs and which leg is this ?
you said you have CUBE in between so you will see 2 seprated H245
negotiation for each leg ,
can you post the H225 and h245 debugs for me please ?
just to make sure that we both talking about the same thing , this
call is Slow start and you have CUBE with Transcoder in it and the
issue you trying to trace is that once you connected the call it got
dropped by the remote GK ?
On the CUBE you have inbound dial-peer with codec G711 and outbound
dial-peer with G729 and then you have transcoder to fix this in the
cube , but on the remote GK you have dial-peer with G711u call only,
if any of the above is not what you have please correct me ,
Ash
On Thu, Dec 1, 2011 at 2:53 PM, ccielabrat ccielab...@gmail.com wrote:
ok, I'm getting to understand this better.
I don't see any mention of a tcs failure though
See the output of debug h245 asn1 below. Where is the indication of a
failure?
Also, I have CUBE running with a Hw transcoder registered locally on HQ
telephony-service.
I would think the CUBE should allocate the xcoder to get around the codec
mismatch.
Output from Debug of H245 ASN1 on HQ/GK/CUBE
Dec 1 20:45:36.806: h245_decode_one_pdu: more_pdus = 0,
bytesLeftToDecode =
97
Dec 1 20:45:36.806: H245 MSC INCOMING ENCODE BUFFER::=
0270010600088175000A801380003C000101010CC0010001000680240001058124080105822280058322C005848501408585011080002B85015000820300010002000301000400052B
Dec 1 20:45:36.806:
Dec 1 20:45:36.806: H245 MSC INCOMING PDU ::=
value MultimediaSystemControlMessage ::= request : terminalCapabilitySet
:
{
sequenceNumber 1
protocolIdentifier { 0 0 8 245 0 10 }
multiplexCapability h2250Capability :
{
maximumAudioDelayJitter 60
receiveMultipointCapability
{
multicastCapability FALSE
multiUniCastConference FALSE
mediaDistributionCapability
{
{
centralizedControl FALSE
distributedControl FALSE
centralizedAudio FALSE
distributedAudio FA
HQ#LSE
centralizedVideo FALSE
distributedVideo FALSE
}
}
}
transmitMultipointCapability
{
multicastCapability FALSE
multiUniCastConference FALSE
mediaDistributionCapability
{
{
centralizedControl FALSE
distributedControl FALSE
centralizedAudio FALSE
distributedAudio FALSE
centralizedVideo FALSE
distributedVideo FALSE
}
}
}
receiveAndTransmitMultipointCapability
{
multicastCapability FALSE
multiUniCastConference FALSE
mediaDistributionCapability
{
{
centralizedControl FALSE
distributedControl FALSE
centralizedAudio FALSE
distributedAudio FALSE
centralizedVideo FALSE
distributedVideo FALSE
[OSL | CCIE_Voice] WAN QOS Strategy Question.
To All, I've been trying to figure the best/fastest way to get a WAN QOS requirement completed on exam day. I've become very comfortable with Auto-QOS and making the needed tweaks, so Auto-QOS is the way I'm going to use. The one piece of the strategy that I'm stilll wondering about is if WAN QOS is specified for only one of the PVC's. Auto-QOS will automatically put Frame-relay traffic shaping on the physical interface which has the side effect of leaving the other pvc with a 56k PVC speed. My solution here is to create a frame-relay map-class with the following parameters. map-class frame-relay Not56k frame-relay traffice-rate 1536 I apply this map-class to the other sub-interface/PVC which negates the 56k problem. I'm curious if anyone has an opinion on the Pros/Cons of this approach and if it might negate requirements somehow. ___ 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] Adding second language to CUE
Hi Ashraf, See below. Thank you! ftp ls 200 Port command successful 150 Opening data channel for directory list. cue-installer.nm-aim.7.0.1 cue-installer.nm-aim.7.0.6 cue-vm-en_GB-langpack.nm-aim.7.0.6.prt1 cue-vm-full-k9.nm-aim.7.0.1.prt1 cue-vm-full-k9.nm-aim.7.0.6.prt1 cue-vm-installer-k9.nm-aim.7.0.1.prt1 cue-vm-installer-k9.nm-aim.7.0.6.prt1 cue-vm-k9.nm-aim.7.0.1.pkg cue-vm-k9.nm-aim.7.0.1.zip cue-vm-k9.nm-aim.7.0.6.pkg cue-vm-k9.nm-aim.7.0.6.zip cue-vm-k9.nmx.7.1.2.zip cue-vm-langpack.nm-aim.7.0.1.pkg cue-vm-langpack.nm-aim.7.0.6.pkg cue-vm-license_12mbx_ccm_7.0.1.pkg cue-vm-license_12mbx_ccm_7.0.6.pkg cue-vm-license_12mbx_cme_7.0.1.pkg cue-vm-license_12mbx_cme_7.0.6.pkg On Thu, Dec 1, 2011 at 2:57 AM, Ashraf Ayyash ash.ayy...@gmail.com wrote: This mean you are installing the Wrong Language files , or you missing on critical file , can you please paste what you have in the FTP directory root ? Ash On Wed, Nov 30, 2011 at 1:03 PM, ccielabrat ccielab...@gmail.com wrote: I'm trying to add a second language to an AIM-CUE. I use the command software install add url ftp://x.x.x.x/xyz.pkg and it seems to run without a problem but when it finishes processing the file, I get the follow message : Language add-ons found on the system (1): Installed SKUName (version) -- * ENU CUE Voicemail US English (7.0.6) Maximum allowed language add-ons (=1) already installed. You can use software uninstall to remove add-ons. ui_install scripts executed successfully. The issue is if I run Show software licenses , it indicates a max of 2 languages are allowed. CUE# sho software licenses Installed license files: - voicemail_lic.sig : 12 MAILBOX LICENSE Core: - Application mode: CCME - Total usable system ports: 6 Voicemail/Auto Attendant: - Max system mailbox capacity time: 840 - Default # of general delivery mailboxes: 5 - Default # of personal mailboxes: 12 - Max # of configurable mailboxes: 17 Interactive Voice Response: - Max # of IVR sessions: Not Available Languages: - Max installed languages: 2 - Max enabled languages: 2 ___ 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 http://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] Adding second language to CUE
oh, ok. I'll give it a try. On Thu, Dec 1, 2011 at 1:44 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote: Hello , why you have tow packages in the root directory ? you have to have the full package of 7.0.6 and the lang pack of GB 7.0.6 ONLY on the root directory , run the installation again and see how it will go Ash On Thu, Dec 1, 2011 at 8:02 AM, ccielabrat ccielab...@gmail.com wrote: Hi Ashraf, See below. Thank you! ftp ls 200 Port command successful 150 Opening data channel for directory list. cue-installer.nm-aim.7.0.1 cue-installer.nm-aim.7.0.6 cue-vm-en_GB-langpack.nm-aim.7.0.6.prt1 cue-vm-full-k9.nm-aim.7.0.1.prt1 cue-vm-full-k9.nm-aim.7.0.6.prt1 cue-vm-installer-k9.nm-aim.7.0.1.prt1 cue-vm-installer-k9.nm-aim.7.0.6.prt1 cue-vm-k9.nm-aim.7.0.1.pkg cue-vm-k9.nm-aim.7.0.1.zip cue-vm-k9.nm-aim.7.0.6.pkg cue-vm-k9.nm-aim.7.0.6.zip cue-vm-k9.nmx.7.1.2.zip cue-vm-langpack.nm-aim.7.0.1.pkg cue-vm-langpack.nm-aim.7.0.6.pkg cue-vm-license_12mbx_ccm_7.0.1.pkg cue-vm-license_12mbx_ccm_7.0.6.pkg cue-vm-license_12mbx_cme_7.0.1.pkg cue-vm-license_12mbx_cme_7.0.6.pkg On Thu, Dec 1, 2011 at 2:57 AM, Ashraf Ayyash ash.ayy...@gmail.com wrote: This mean you are installing the Wrong Language files , or you missing on critical file , can you please paste what you have in the FTP directory root ? Ash On Wed, Nov 30, 2011 at 1:03 PM, ccielabrat ccielab...@gmail.com wrote: I'm trying to add a second language to an AIM-CUE. I use the command software install add url ftp://x.x.x.x/xyz.pkg and it seems to run without a problem but when it finishes processing the file, I get the follow message : Language add-ons found on the system (1): Installed SKU Name (version) -- * ENU CUE Voicemail US English (7.0.6) Maximum allowed language add-ons (=1) already installed. You can use software uninstall to remove add-ons. ui_install scripts executed successfully. The issue is if I run Show software licenses , it indicates a max of 2 languages are allowed. CUE# sho software licenses Installed license files: - voicemail_lic.sig : 12 MAILBOX LICENSE Core: - Application mode: CCME - Total usable system ports: 6 Voicemail/Auto Attendant: - Max system mailbox capacity time: 840 - Default # of general delivery mailboxes: 5 - Default # of personal mailboxes: 12 - Max # of configurable mailboxes: 17 Interactive Voice Response: - Max # of IVR sessions: Not Available Languages: - Max installed languages: 2 - Max enabled languages: 2 ___ 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] Need some help with GK/CUBE Remote Zone call debugging.
ok, I'm getting to understand this better.
I don't see any mention of a tcs failure though
See the output of debug h245 asn1 below. Where is the indication of a
failure?
Also, I have CUBE running with a Hw transcoder registered locally on HQ
telephony-service.
I would think the CUBE should allocate the xcoder to get around the codec
mismatch.
Output from Debug of H245 ASN1 on HQ/GK/CUBE
Dec 1 20:45:36.806: h245_decode_one_pdu: more_pdus = 0, bytesLeftToDecode
= 97
Dec 1 20:45:36.806: H245 MSC INCOMING ENCODE BUFFER::=
0270010600088175000A801380003C000101010CC0010001000680240001058124080105822280058322C005848501408585011080002B85015000820300010002000301000400052B
Dec 1 20:45:36.806:
Dec 1 20:45:36.806: H245 MSC INCOMING PDU ::=
value MultimediaSystemControlMessage ::= request : terminalCapabilitySet :
{
sequenceNumber 1
protocolIdentifier { 0 0 8 245 0 10 }
multiplexCapability h2250Capability :
{
maximumAudioDelayJitter 60
receiveMultipointCapability
{
multicastCapability FALSE
multiUniCastConference FALSE
mediaDistributionCapability
{
{
centralizedControl FALSE
distributedControl FALSE
centralizedAudio FALSE
distributedAudio FA
HQ#LSE
centralizedVideo FALSE
distributedVideo FALSE
}
}
}
transmitMultipointCapability
{
multicastCapability FALSE
multiUniCastConference FALSE
mediaDistributionCapability
{
{
centralizedControl FALSE
distributedControl FALSE
centralizedAudio FALSE
distributedAudio FALSE
centralizedVideo FALSE
distributedVideo FALSE
}
}
}
receiveAndTransmitMultipointCapability
{
multicastCapability FALSE
multiUniCastConference FALSE
mediaDistributionCapability
{
{
centralizedControl FALSE
distributedControl FALSE
centralizedAudio FALSE
distributedAudio FALSE
centralizedVideo FALSE
distributedVideo FALSE
}
}
}
mcCapability
{
centralizedConferenceMC FALSE
decentralizedConferenceMC FALSE
}
rtcpVideoControlCapability FALSE
mediaPacketizationCapability
{
h261aVideoPacketization FALSE
}
logicalChannelSwitchingCapability FALSE
t120DynamicPortCapability FALSE
}
capabilityTable
{
{
capabilityTableEntryNumber 1
capability receiveAudioCapability : g729wAnnexB : 6
},
{
capabilityTableEntryNumber 2
capability receiveAudioCapability : g729AnnexAwAnnexB : 6
},
{
capabilityTableEntryNumber 3
capability receiveAudioCapability : g729 : 6
},
{
capabilityTableEntryNumber 4
capability receiveAudioCapability : g729AnnexA : 6
},
{
capabilityTableEntryNumber 5
capability receiveAndTransmitUserInputCapability : dtmf : NULL
},
{
capabilityTableEntryNumber 6
capability receiveAndTransmitUserInputCapability : basicString :
NULL
},
{
capabilityTableEntryNumber 44
capability receiveAndTransmitUserInputCapability : hookflash :
NULL
}
}
capabilityDescriptors
{
{
capabilityDescriptorNumber 0
simultaneousCapabilities
{
{
1,
2,
3,
4
},
{
5,
6
},
{
44
}
}
}
}
}
Dec 1 20:45:36.810: h245_decode_one_pdu: H245ASNDecodePdu rc = 0,
bytesLeftToDecode = 0
Dec 1 20:45:36.810: h245_decode_one_pdu: Read Pkt body: more_pdus:0 rc:0
asn_rc:0
HQ#
HQ#
HQ#
HQ#sho deb
H.245:
H.245 ASN1 Messages debugging is on
On Thu, Dec 1, 2011 at 2:00 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote:
The ccapi debug will show you the cause code which doesn't explain why
the call failed ,
you have to debug the h245 asn1 and check the TCS and see the codecs
advertised and received and then you will get the TCS negotiation
failure so you can explain that there is codec mismatch
Ash
On Thu, Dec 1, 2011 at 11:55 AM, Mohd Baqari baqari.voic...@gmail.com
wrote:
Use the command debug voice ccapi inout. H323 debugs won't show in this
case.
Regards,
Mohammed Al Baqari
Sent from my iPhone
On Dec 1, 2011, at 6:12 PM, ccielabrat ccielab...@gmail.com wrote:
I'm trying to setup a call from HQ
[OSL | CCIE_Voice] Creating log files in the lab.
I don't think this is an NDA topic. If you think it is, please disregard. Is there any restriction to create log files on the Lab pc you work off in the lab? I like to grab the default CUE config into a text file and modify then paste back. Just wondering if the PC is locked down that I can't save files along the way. ___ 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] Adding second language to CUE
I'm trying to add a second language to an AIM-CUE. I use the command software install add url ftp://x.x.x.x/xyz.pkg and it seems to run without a problem but when it finishes processing the file, I get the follow message : Language add-ons found on the system (1): Installed SKUName (version) -- * ENU CUE Voicemail US English (7.0.6) Maximum allowed language add-ons (=1) already installed. You can use software uninstall to remove add-ons. ui_install scripts executed successfully. The issue is if I run Show software licenses , it indicates a max of 2 languages are allowed. CUE# sho software licenses Installed license files: - voicemail_lic.sig : 12 MAILBOX LICENSE Core: - Application mode: CCME - Total usable system ports: 6 Voicemail/Auto Attendant: - Max system mailbox capacity time: 840 - Default # of general delivery mailboxes: 5 - Default # of personal mailboxes: 12 - Max # of configurable mailboxes: 17 Interactive Voice Response: - Max # of IVR sessions: Not Available Languages: - Max installed languages: 2 - Max enabled languages: 2 ___ 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 with H323 GW
The problem was the check box on the gateway configuration. After looking at every available option on the gateway, I found out CUCM wasn't send the name in the first place :) I do have a follow up question though. I'm really confused about the logic around no supplementary-service h225-notify cid-update If the command is by default supplementary-service h225-notify cid-update, it seems to me this would ENABLE the CID to be updated. It doesn't make sense to me why DISABLING this with NO supplementary-service h225-notify cid-update actually allows the gateway to trigger an updated display on the phone. On Tue, Nov 29, 2011 at 5:06 AM, datucha123 datucha123 datucha...@gmail.com wrote: Also you have to check the Display IE checkbox in CUCM H323 gateway configuration. On Tue, Nov 29, 2011 at 7:05 AM, Rrcrumm rrcr...@yahoo.com wrote: Do you have isdn outgoing display-ie under serial 0/0/0:23 or 15 interface? Randall Sent from my iPhone On Nov 28, 2011, at 5:52 PM, ccielabrat ccielab...@gmail.com wrote: I must be missing something easy. I'm trying to get Calling Name to display on my PSTN phone when receiving a call from a IP phone going through a H323 gateway. I've found many links online suggesting it's not supported but then others suggesting it's possible. Can someone point me to a good link? ___ 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 http://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 http://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] Calling Name with H323 GW
I must be missing something easy. I'm trying to get Calling Name to display on my PSTN phone when receiving a call from a IP phone going through a H323 gateway. I've found many links online suggesting it's not supported but then others suggesting it's possible. Can someone point me to a good link? ___ 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] Question about Called Number displayed on phone.
Can someone help me understand what determines what gets displayed on the phone display when calling outbound. I have a setup where I have a h323 Gw and MGCP Gw in a single RL. I create a route pattern of 9.2345678 and assign it to the RL. If it goes to the H323 GW , I don't drop the 9 prefix in the RL and it displays 92345678 on the phone. If it goes to the MGCP GW, the 9 prefix is dropped in the RL and it displays 2345678 on the phone. So I figured the display value must be based on what gets sent to the GW, but this doesn't seem to be true either. If I adjust my dial-peers on H323 to match on 2345678 (no 9 prefix) , and drop the 9 in the RL , I still see the 9 prefix as dialed on the phone. ___ 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] TFTP Question
Group, Is there a service parameter to set for TFTP on CUCM to allow the TFTP server to see new files uploaded without restarting the TFTP Service ? ___ 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] Question about CUBE Gatekeepers
I need clarification about Gatekeepers using outvia to a CUBE zone. I've always thought a CUBE config needed the underlying Telephony-service config to be operational. Is that the case? I suppose if the call setup is using g.729 in/out of the CUBE , there is no need to have anything but a matching dial-peer and the allow h323 to h323 in the voice service voip. Can someone confirm or correct my understanding. 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] cue-cme mwi unsolicited question
Hey Ashraf, You got me thinking the right way. I had a mismatch between my sip interface and the gateway configured on CUE. Thanks! On Sat, Oct 22, 2011 at 4:43 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote: did you binded the SIP to the correct interface from the CME config Voice service Voip ? Any chance to reload the Funky CUE ? Ash On Sat, Oct 22, 2011 at 12:12 PM, ccielabrat ccielab...@gmail.com wrote: I can't get CUE MWI working either. This is my cue config for SIP ccn subsystem sip gateway address 10.1.131.1 mwi envelope-info mwi sip unsolicited end subsystem I've tried all kinds of config on the CME router without success. When running debug ccsip messages on the CME router , I don't see anything if I issue mwi refresh all on CUE, even though I can dial into CUE and check to hear a voicemail on dn 4001 On Fri, Oct 21, 2011 at 6:47 PM, Brian btmulg...@gmail.com wrote: hi - this is an excellent summary of mwi for cue that is worth a read http://blog.ipexpert.com/2010/07/19/sip-mwi-mechansims-on-cue-notify/ Sent from my iPad On 21 Oct 2011, at 21:23, Ashraf Ayyash ash.ayy...@gmail.com wrote: Hello Zamuel , the mwi relay command is only needed in case of the subscribe notify MWI and its not needed in case of using Unsolicited because it does send the event to the phone using NOTIFY message no matter it subscribed to the MWI server Or not . Ash On Fri, Oct 21, 2011 at 11:21 AM, zamuel del Toro sdelto...@hotmail.com wrote: Hi Vic, how is it going?, about mwi unsolicited. sip-ua mwi.. unsolicited telephony-ser mwi relay ephone-dn nothing works mwi if subscribe notify sip-ua mwi... telephony-ser nothing ephone-dn mwi sip both works fine what if make mistake if on unsolicited include on ephone-dn mwi sip, that work too.is wrong do this? 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 ___ 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-cme mwi unsolicited question
I can't get CUE MWI working either. This is my cue config for SIP ccn subsystem sip gateway address 10.1.131.1 mwi envelope-info mwi sip unsolicited end subsystem I've tried all kinds of config on the CME router without success. When running debug ccsip messages on the CME router , I don't see anything if I issue mwi refresh all on CUE, even though I can dial into CUE and check to hear a voicemail on dn 4001 On Fri, Oct 21, 2011 at 6:47 PM, Brian btmulg...@gmail.com wrote: hi - this is an excellent summary of mwi for cue that is worth a read http://blog.ipexpert.com/2010/07/19/sip-mwi-mechansims-on-cue-notify/ Sent from my iPad On 21 Oct 2011, at 21:23, Ashraf Ayyash ash.ayy...@gmail.com wrote: Hello Zamuel , the mwi relay command is only needed in case of the subscribe notify MWI and its not needed in case of using Unsolicited because it does send the event to the phone using NOTIFY message no matter it subscribed to the MWI server Or not . Ash On Fri, Oct 21, 2011 at 11:21 AM, zamuel del Toro sdelto...@hotmail.com wrote: Hi Vic, how is it going?, about mwi unsolicited. sip-ua mwi.. unsolicited telephony-ser mwi relay ephone-dn nothing works mwi if subscribe notify sip-ua mwi... telephony-ser nothing ephone-dn mwi sip both works fine what if make mistake if on unsolicited include on ephone-dn mwi sip, that work too.is wrong do this? 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 http://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 http://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 http://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] Resend: Task Order in the lab
No takers? -- Forwarded message -- From: ccielabrat ccielab...@gmail.com Date: Sat, Sep 17, 2011 at 12:11 PM Subject: Task Order in the lab To: ccie_voice@onlinestudylist.com To All, I know this question has been asked 1000 times. Now that I am ready to schedule my lab exam, I need to ask it again. How are successful exam takers breaking down and grouping the tasks that make up the lab to allow for time savings ? My natural approach so far is to take a sheet of paper and divide it into 4 boxes (HQ, BR1, BR2, CUCM) Then I read through the practice exam and note the task # (i.e. 1.1 ) and a 1-3 word description in the device box that it relates to. If a task includes or impacts multiple devices, it goes in multiple boxes. I also note DN's in each location box and call restrictions for each dn in shorthand. In the CUCM box, I keep an area that notes service parameters that need attention , in addition to the more general stuff. In trying this a couple times, it seems I can get a straw man topology working in short order. But I'm concerned that it leaves me in a position where I will end up jumping around to fine tune devices/locations multiple times. I know the common strategy that was used in the prior version of the lab (Touch each device once) is generally not the way with the new format. Maybe with the troubleshooting in the exam now, my approach is flawed. Any opinion or insight is appreciated. LabRat ___ 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] Task Order in the lab
To All, I know this question has been asked 1000 times. Now that I am ready to schedule my lab exam, I need to ask it again. How are successful exam takers breaking down and grouping the tasks that make up the lab to allow for time savings ? My natural approach so far is to take a sheet of paper and divide it into 4 boxes (HQ, BR1, BR2, CUCM) Then I read through the practice exam and note the task # (i.e. 1.1 ) and a 1-3 word description in the device box that it relates to. If a task includes or impacts multiple devices, it goes in multiple boxes. I also note DN's in each location box and call restrictions for each dn in shorthand. In the CUCM box, I keep an area that notes service parameters that need attention , in addition to the more general stuff. In trying this a couple times, it seems I can get a straw man topology working in short order. But I'm concerned that it leaves me in a position where I will end up jumping around to fine tune devices/locations multiple times. I know the common strategy that was used in the prior version of the lab (Touch each device once) is generally not the way with the new format. Maybe with the troubleshooting in the exam now, my approach is flawed. Any opinion or insight is appreciated. LabRat ___ 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] UnifiedFX phoneview question
Anyone successfully get Phoneview working with CUCME? I see express as an option in the group configuration with telnet:// as the protocol to use but I don't have any idea what would be needed in the CUCME config. Thanks LabRat ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Couple questions about MOH on CUCME
Hi All, I'm working on a better understanding of the options with MOH from CME. I have one endpoint registered as SCCP and one as SIP. Questions: 1.) Does the SIP phone configuration for MOH get defined in the telephony-service area? 2.) And does multicast MOH to sip endpoints work the same as SCCP ? 3.) If multicast moh is configured , is there any option for the phone to fallback to unicast if it can't join the multicast? What I have now is a call from SIP to SCCP works and if the SIP phone holds the call, I do get moh on the SCCP. If the SCCP phone holds the call, I get beeps on the SIP phone. Any insight is appreciated. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] OT : Missing normal softkeys from 9971.
I'm hoping i missed something simple. I just got a 9971 registered on a CUCM 7.x server. it works great but I noticed there are no available softkeys for hold, park , etc during a call. It's my first SIP phone I'm using, so is there something I'm missing regarding supplemental services? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] OT : Missing normal softkeys from 9971.
Right, thats why I included OT : (Off topic) in the subject line. That way, for people who don't want to be bothered can simply ignore it. On Wed, Mar 16, 2011 at 4:14 PM, Roger Carpio roger.car...@gmail.comwrote: 9971 are not evaluated in CCIE Lab as far as I know therefore, this is not the right place for this question. Try doing some google or post this question at Cisco Support forums. http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/9971_9951_8961/8_5/english/user_guide/book_911/callingfeatures_rt.html#wp1027556 Regards, Roger Carpio. On Wed, Mar 16, 2011 at 1:41 PM, ccielab...@gmail.com wrote: I'm hoping i missed something simple. I just got a 9971 registered on a CUCM 7.x server. it works great but I noticed there are no available softkeys for hold, park , etc during a call. It's my first SIP phone I'm using, so is there something I'm missing regarding supplemental services? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] OT : Missing normal softkeys from 9971.
Randall Roger, Are you kidding me? I admit it was a dumb question and I tried to be polite by marking it OT: (Off Topic) I was stuck in a lab with no direct internet access or a manual for a phone I never used before. I leveraged the mailer group on my phone looking for a hand. A group, which for the most part, is helpful whether or not it has anything specifically to do with the voice lab. I got several quick replies with useful information. The thread could have died 5 minutes after I posted it. In the time it took to reply, both of you could have deleted it and moved on with your day. I don't see either of you replying back to morons who blatantly ask for NDA information. *Your arrogance is laughable*, to think either of you have a place to suggest what does or doesn't belong on this mailing list. If you have a problem with it , take it up with the mailer admin. Or email me a picture of your mailer police badge , and then I'll consider your email. I honestly hope if you are ever in the same position, you both get a helpful reply instead of the nonsense you both provided. On Wed, Mar 16, 2011 at 9:21 PM, Randall Saborío Cubero ill2...@gmail.comwrote: It's difficult to ignore it if it is addressed to a specialized mailing list. It also takes extra time to sort out the crap that does not belong from the important emails. I'm sure you will get some attention at the cisco support forums and 0 rejection. El mié, 16-03-2011 a las 17:35 -0400, ccielabrat escribió: Right, thats why I included OT : (Off topic) in the subject line. That way, for people who don't want to be bothered can simply ignore it. On Wed, Mar 16, 2011 at 4:14 PM, Roger Carpio roger.car...@gmail.com wrote: 9971 are not evaluated in CCIE Lab as far as I know therefore, this is not the right place for this question. Try doing some google or post this question at Cisco Support forums. http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/9971_9951_8961/8_5/english/user_guide/book_911/callingfeatures_rt.html#wp1027556 Regards, Roger Carpio. On Wed, Mar 16, 2011 at 1:41 PM, ccielab...@gmail.com wrote: I'm hoping i missed something simple. I just got a 9971 registered on a CUCM 7.x server. it works great but I noticed there are no available softkeys for hold, park , etc during a call. It's my first SIP phone I'm using, so is there something I'm missing regarding supplemental services? ___ 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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Converting SCCP phones to SIP
Looks like I'm right back in the same mess again. Now I can't seem to get the CIPC currently running SIP to run as a SCCP phone. I even changed the TFTP away from CME and pointed it to CUCM thinking that would force the phone to register as an SCCP phone. No good. Is there a factory reset' type action available on CIPC ? Looking at the logs , phone is simply requesting the SIP config file from CUCM and failing with an unprovisioned message on the phone. I've tried to configure the CIPC endpoint as SCCP and SIP without getting either to work. I'm in the weeds without a plan. If anyone can point me in any direction, I'd appreciate it . On Wed, Aug 11, 2010 at 7:55 PM, Cisco CCIE ccieforl...@gmail.com wrote: Great feeling when you figure it out on your own I figured out the problem. And as usual, I'm the architect of my own demise :) I had redundant tftp-server statements for the term62.default.loads file. (along with others as well) One for SCCP and one for SIP. tftp-server flash:/SCCP.8-3-3s/term62.default.loads alias term62.default.loads tftp-server flash:/SIP.8-3-3/term62.default.loads alias term62.default.loads Apparently, if there are redundant tftp-server commands, the first one (in this case SCCP) takes precedence. As soon as I removed the first statement pointing towards the SCCP term62.default.loads, the phone took the SIP image. One more problem solved by endless hours of frustration ... ___ 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
[OSL | CCIE_Voice] RTMT for Dummies
Can anyone point me towards a good RTMT link that will give the basics of how to collect a trace from CUCM 7.x ? I'm looking into a GK related issue and what to see what messaging is happening on the CUCM side. I'm stumbling badly with RTMT. Any help or links are greatly appreciated. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Converting SCCP phones to SIP
I figured out the problem. And as usual, I'm the architect of my own demise :) I had redundant tftp-server statements for the term62.default.loads file. (along with others as well) One for SCCP and one for SIP. tftp-server flash:/SCCP.8-3-3s/term62.default.loads alias term62.default.loads tftp-server flash:/SIP.8-3-3/term62.default.loads alias term62.default.loads Apparently, if there are redundant tftp-server commands, the first one (in this case SCCP) takes precedence. As soon as I removed the first statement pointing towards the SCCP term62.default.loads, the phone took the SIP image. One more problem solved by endless hours of frustration ... On Sat, Jul 31, 2010 at 7:09 PM, ccielabrat ccielab...@gmail.com wrote: Maybe I'm missing a part of this process. Should a factory reset be necessary? On 7/31/10, Miron Kobelski findko...@gmail.com wrote: You have your load command in voice register global? Restore the phones to factory defaults and provide on the TFTP all the files it will ask for. It always worked for me. But I know longer change firmware on CUCME itself - it's so much easier and quicker by temporary registering the phone on CUCM. regards kobel On Sun, Aug 1, 2010 at 12:13 AM, ccielabrat ccielab...@gmail.com wrote: ete the ephone config (sccp) - issue no create cnf (to delete the sccp cnf info) - Created voice register pool with the mac address of the phone. - Loaded the sip firmware on the router and made files available via tftp-server command. No matter what I do, the thing will NOT ask for the SIP firmware. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Converting SCCP phones to SIP
I'm running into a strange problem where an existing phone running sccp firmware will not register using SIP firmware. I've done the following - Delete the ephone config (sccp) - issue no create cnf (to delete the sccp cnf info) - Created voice register pool with the mac address of the phone. - Loaded the sip firmware on the router and made files available via tftp-server command. No matter what I do, the thing will NOT ask for the SIP firmware. I'm using 8.3.3 firmware for SCCP and SIP. Thanks in advance. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Converting SCCP phones to SIP
Maybe I'm missing a part of this process. Should a factory reset be necessary? On 7/31/10, Miron Kobelski findko...@gmail.com wrote: You have your load command in voice register global? Restore the phones to factory defaults and provide on the TFTP all the files it will ask for. It always worked for me. But I know longer change firmware on CUCME itself - it's so much easier and quicker by temporary registering the phone on CUCM. regards kobel On Sun, Aug 1, 2010 at 12:13 AM, ccielabrat ccielab...@gmail.com wrote: ete the ephone config (sccp) - issue no create cnf (to delete the sccp cnf info) - Created voice register pool with the mac address of the phone. - Loaded the sip firmware on the router and made files available via tftp-server command. No matter what I do, the thing will NOT ask for the SIP firmware. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Question on Alias Static configuration.
Assuming the following requirements for BR2 to HQ calling via Gatekeeper: - Do not use tech prefix or default tech prefix - Route calls to Sub and then to Pub if Sub unavailable. So in this case a Static Alias configuration would be needed to allow the call to route to CUCM without having the usual 1# tech prefix. The problem I'm seeing is you can only target a single IP for any given e164 number. Is it possible to use static alias and still allow for redundancy ? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] H323 gateway inbound calls issue.
Maybe a codec selection issue. Do you have any restrictions on CUCM (Regions, Locations) that would disallow the call setup? On Sat, Jun 12, 2010 at 11:03 AM, jammer jones jammerjone...@gmail.comwrote: What would cause the following issues with a gw configured as h323 to cucm outbound calls work just fine inbound calls from a pstn phone does not work. PSTN phone displays unknown number and then a fast busy - this is the problem. when in srst mode outbound and inbound calls work just fine. - this makes me believe it is something in cucm. I verified the config on the gateway in cucm 6 times. My config on the gateway was this. voice translation-rule 1 rule 1 /.*\(\)/ /\1/ dial-peer voice 1 pots translation-profile incoming frompstn incoming called-number . direct-inward-dial port 0/0/0:23 card type t1 0 0 network-clock participate wic 0 isdn switch-type primary-ni controller t1 0/0/0 pri-group timeslots 1-24 inter ser 0/0/0:23 isdn outgoing display-ie isdn outgoing ie redirecting-number interface vlan 302 h323-gateway voip bind srcaddr 1.1.1.1 dial-peer voice 100 voip destination-pattern 3...$ session target ipv4:2.2.2.2 dtmf-relay h245-alphanumeric no vad dial-peer voice 101 voip preference 1 destination-pattern 7...$ session target ipv4:2.2.2.1 dtmf-relay h245-alphanumeric no vad config on cucm was all of the usual. I verified i was pointing to 1.1.1.1 several times, verified i had a correct inbound css on gateway, etc.. the debug isdn q931 showed invalid information element. the debug cch323 h225 showed what appeared to be a proper call flow. Unfortunately i am unable to reproduce the same behavior so i do not have better debug messages. could something me set coming in from the telco that would break inbound calls going to phones registered to cucm, but not phones registered to srst? Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Expensive Lunch
Just my 2 cents. I had a very similar experience as you in terms of scoring. Some items I know where working got no points. It's my impression (and others I've spoken with) is that once the proctor gets to -21 on your score, they simply stop any further testing in most cases. I could be wrong and I would suspect they are instructed by Cisco to score the whole exam but I don't believe they do. On Fri, May 21, 2010 at 2:25 PM, Ashar Siddiqui siddas...@gmail.com wrote: Don't worry Jeff, I am also in the same boat as you. Few questions where I was sure to get 100%, I didn't get! You will have to sit and carefully look at what was asked and what you configured. Sometimes getting desired results is not the solution, getting desired results as per Cisco way is what they are looking for. Ash Kevin Damisch wrote: I feel your pain. I remember not getting points for a section that I triple checked from each phone at the end of the day, ran the debugs to make sure number/type were set to what is asked, checked the display on both the calling/called phones, all matched up to what was asked as it was slightly different than what you would expect due to the curveballs they like to throw at you. Very tough to know what was missed in these situations. Make sure you document everything, go home, and mock it up again. I've caught myself on a few things by doing that. Best of luck next time. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Cotter [jcot...@voxns.com] Sent: Friday, May 21, 2010 11:59 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Expensive Lunch Sat the lab yesterday….failed. So I need to vent a little bit. There is no doubt I failed the exam and it was painfully obvious I am not ready to be a CCIE. However the most unsettling piece is not getting the points in areas that I “thought” were working and verified. I do not know how to address this the next time. My fear is, I will probably program these items exactly the same way next time… because that is the way I know how….they coincide with the training materials available and most importantly they seem to work…. and I will not get the points AGAIN!! I just do not understand why I did not get these points or how to fix it the next time….frustrated. Lunch was good though! This communication (including any attachments) is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If you are not the intended recipient, any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify Vital Support Systems at 515 334 5700 and delete or destroy all copies and the original document. ___ 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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Console cable for a 7970
I've found a couple links about being able to get a console session to an IP phone via the AUX port.Completely unsupported but apparently do-able. Has anyone messed with this? I have a 7970 that refuses to boot, it flashes but then goes slient. Before I trash it, I figured I would try to find out if it posts an error through this interface. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] DSCP for SCCP
In terms of modifying the signaling DSCP value for phones and what is generated by CallManager, Are the entries noted below the only ones I need to check? Enterprise Parameters - DSCP for Cisco CallManager to Device Interface* - DSCP for SCCP Phone Configuration* Service Parameters: - Voice Streaming APP: IP Type of Service to Cisco CallManager* - CTI Manager Services Advanced: DSCP for ICCP Protocol Links* DSCP IP CTIManager to Application* ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Clarifying VPM Broadcast setup
I think I have the VPIM Broadcast config straight now.But I want to confirm I'm doing it the normal way. CUE is configured with it's location VPIM-Broadcast ID as 852. I'm creating a VPIM subscriber with an remote mailbox of 852 and assigning it to the cue vpim location. I'm creating a PDL and including all HQ and SB phones and VPIM user 852 in it. From Unity Broadcast admin , i'm selecting as a PDL and leaving the message. What I'm noticing is that the message is delivered as a normal VM to the HQ and SB sites but as a true broadcast to the CUE users. It seems to work (MWI, etc.) , just curious if there is a variant on the config . Also, I'm assuming the same basic scenario is used for CUE to Unity Broadcast ? But Unity doesn't light an MWI for broadcast delivery right ?
[OSL | CCIE_Voice] Follow up on an old question (Unity MWI)
I just discovered a behavior that surprised me. If I register only 3 voicemail ports in CM but have 8 port configured in Unity, the mwi's behavior is unpredictable. I can only guess that Unity is attempting to use port 4-8 for MWI , regardless of there status within the system. As soon as I uncheck the enabled button in the UTIM for ports 4-8, MWI starts working great. Just something to watch out for. - Scott
[OSL | CCIE_Voice] Voice translation question.
I'd like to be able to match a dial string with the format *400.I'd like to match the beginning * and then strip it. I can't seem to get a valid voice translation rule though. I assumed I would have to escape the * by using \* but I get an error when I try to use the following. rule 1 /^\*\(400.\)/ /\1/ I'm I thinking about this incorrectly? I also tried just having * as a single character (no escape). The documentation suggests that I should be able to do this. But I get an error with this one also. rule 1 /*\(400.\)/ /\1/
Re: [OSL | CCIE_Voice] Voice translation question.
Thanks for the quick reply. Actually it is working , I fat fingered it in the config. On Wed, Jun 24, 2009 at 3:30 PM, Cyrus cyrus@gmail.com wrote: Hi, It's not work with translation-rule 1 use voice translation-rule 1 rule 1 /^\*\(400.\)/ /\1/ Cheers, Cyrus On Thu, Jun 25, 2009 at 5:18 AM, ccielab...@gmail.com wrote: I'd like to be able to match a dial string with the format *400. I'd like to match the beginning * and then strip it. I can't seem to get a valid voice translation rule though. I assumed I would have to escape the * by using \* but I get an error when I try to use the following. rule 1 /^\*\(400.\)/ /\1/ I'm I thinking about this incorrectly? I also tried just having * as a single character (no escape). The documentation suggests that I should be able to do this. But I get an error with this one also. rule 1 /*\(400.\)/ /\1/ -- Sirus Moghadasian CCIE #21862 (RS)
[OSL | CCIE_Voice] What is the difference between VPIM delivery ?
I'm testing VPIM delivery from Unity to CUE using a PDL with an extension of 666. If I send a vm directly to PDL ext 666 , the message goes through and the MWI light goes on for CUE extensions. If I send a vm via Broadcast Manager to PDL 666 , the message doesn't go through and the MWI light doesn't go on (obviously). The only difference I can see is the subject line in the VPIM trace on CUE reads Broadcast Message instead of From x...@xxx.xxx ?
Re: [OSL | CCIE_Voice] CME Fast Transfer Problem
The monitored line is a ephone-dn of 54xxx. It's not assigned as a standard line to any phone, so it's ALWAYS idle. So , by pressing the transfer softkey and then the button assigned as monitor like it acts as a transfer + speed dial. Strange thing is I wasn't able to get this to work with an actual speed-dial configured on the transferring phone. I would think that it should work the same way but it doesn't. On Wed, Apr 22, 2009 at 8:32 AM, Linda Mordosky (lmordosk) lmord...@cisco.com wrote: Out of curiosity, how are you getting the transfer to go to VM when the monitored line is not busy? The System Guide states: When a monitored line is idle, pressing the monitor button will speed-dial the monitored line. This functionality is sometimes known as fast transfer or direct station select. -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] On Behalf Of Prabahar M Sent: Wednesday, April 22, 2009 12:55 AM To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME Fast Transfer Problem transfer-system full-blind will work for this problem. Is there anyway just to press transfer softkey without answering the call and the monitor button to send call to greeting? There is no transfer sofkey available on the softkey template for alert. Callers hears music before call connects to the greeting after pressing transfer softkey. Thanks, Prabahar On Tue, Apr 21, 2009 at 10:02 PM, ccie_voice-requ...@onlinestudylist.com wrote: Date: Tue, 21 Apr 2009 22:07:44 -0600 From: J Delgado elpela...@hotmail.com Subject: [OSL | CCIE_Voice] CME Fast Transfer Problem To: ccielab...@gmail.com Cc: ccie_voice@onlinestudylist.com I had the same issue. I recommend you to configure under telephony-service transfer-system full-blind so when you transfer the call to Unity, you only need to press the Transfer button once. Cheers, Juan From: ccie_voice-requ...@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 38, Issue 127 To: ccie_voice@onlinestudylist.com Date: Wed, 22 Apr 2009 00:35:28 -0400 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: AAR and Offnet Transfers (James Key) 2. Re: Version 3 Lab equipment... (Michael Ciarfello) 3. CME Fast Transfer Problem. (ccielab...@gmail.com) 4. Re: CME Fast Transfer Problem. (Cliff McGlamry) 5. Re: CME Fast Transfer Problem. (Cliff McGlamry) -- Message: 1 Date: Tue, 21 Apr 2009 19:50:42 -0500 From: James Key j...@jackhenry.com Subject: Re: [OSL | CCIE_Voice] AAR and Offnet Transfers To: Prabahar M prabaha...@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Message-ID: 7105bfd589769844b43b67bbfa844a9b015957c...@mmoexchmbs01.jhacorp.com Content-Type: text/plain; charset=us-ascii Hi Prabahar. It is still not working for me and I can't seem to find a working solution. Has anyone been able to get this to work? James Key From: Prabahar M [prabaha...@gmail.com] Sent: Tuesday, April 21, 2009 5:50 PM To: James Key Subject: Re: AAR and Offnet Transfers Hi James, Does it work for you? I tried and it is not working for me. If it works for you, could you send me the config steps please. Thansk, Prabahar Thanks for the reply Cliff. That is what I was thinking and did configure it this way, but when site B tries to transfer i get the message that transfer is not allowed. I will take a closer look. James Key From: Cliff McGlamry [cliff at mcglamry.net] Sent: Monday, April 20, 2009 7:08 PM To: James Key; ccie_voice at onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] AAR and Offnet Transfers Usually when you get something like this it's because you have a situation where Site A calls site B via AAR. Then the user at site B is transferring the call outside to somewhere else. What you do is set offnet to offnet transfers as disabled. Then you create an AAR route pattern to support the call from A to B, but classify the call as on net. Then the user at B will be able to transfer it even though it would otherwise violate the rule. - Original Message - From: James Keymailto:JKey at jackhenry.com To: ccie_voice at
Re: [OSL | CCIE_Voice] CME Fast Transfer Problem
I haven't tried that.I thought if you have full-consult configured, you would have to press transfer a second time to complete the transfer. I'll test it on my setup which is a 3825 running 12.4.5b On Wed, Apr 22, 2009 at 7:00 PM, Sergio Polizer spoli...@hotmail.comwrote: Have you entered transfer-system full-consult ? I got make this work with same config as you posted (plus the above line) for internal and external calls with IOS 12.4(3g). Sergio. -- Date: Wed, 22 Apr 2009 09:10:04 -0400 From: ccielab...@gmail.com To: lmord...@cisco.com CC: ccie_voice@onlinestudylist.com; prabaha...@gmail.com Subject: Re: [OSL | CCIE_Voice] CME Fast Transfer Problem The monitored line is a ephone-dn of 54xxx. It's not assigned as a standard line to any phone, so it's ALWAYS idle. So , by pressing the transfer softkey and then the button assigned as monitor like it acts as a transfer + speed dial. Strange thing is I wasn't able to get this to work with an actual speed-dial configured on the transferring phone. I would think that it should work the same way but it doesn't. On Wed, Apr 22, 2009 at 8:32 AM, Linda Mordosky (lmordosk) lmord...@cisco.com wrote: Out of curiosity, how are you getting the transfer to go to VM when the monitored line is not busy? The System Guide states: When a monitored line is idle, pressing the monitor button will speed-dial the monitored line. This functionality is sometimes known as fast transfer or direct station select. -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] On Behalf Of Prabahar M Sent: Wednesday, April 22, 2009 12:55 AM To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME Fast Transfer Problem transfer-system full-blind will work for this problem. Is there anyway just to press transfer softkey without answering the call and the monitor button to send call to greeting? There is no transfer sofkey available on the softkey template for alert. Callers hears music before call connects to the greeting after pressing transfer softkey. Thanks, Prabahar On Tue, Apr 21, 2009 at 10:02 PM, ccie_voice-requ...@onlinestudylist.com wrote: Date: Tue, 21 Apr 2009 22:07:44 -0600 From: J Delgado elpela...@hotmail.com Subject: [OSL | CCIE_Voice] CME Fast Transfer Problem To: ccielab...@gmail.com Cc: ccie_voice@onlinestudylist.com I had the same issue. I recommend you to configure under telephony-service transfer-system full-blind so when you transfer the call to Unity, you only need to press the Transfer button once. Cheers, Juan From: ccie_voice-requ...@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 38, Issue 127 To: ccie_voice@onlinestudylist.com Date: Wed, 22 Apr 2009 00:35:28 -0400 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: AAR and Offnet Transfers (James Key) 2. Re: Version 3 Lab equipment... (Michael Ciarfello) 3. CME Fast Transfer Problem. (ccielab...@gmail.com) 4. Re: CME Fast Transfer Problem. (Cliff McGlamry) 5. Re: CME Fast Transfer Problem. (Cliff McGlamry) -- Message: 1 Date: Tue, 21 Apr 2009 19:50:42 -0500 From: James Key j...@jackhenry.com Subject: Re: [OSL | CCIE_Voice] AAR and Offnet Transfers To: Prabahar M prabaha...@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Message-ID: 7105bfd589769844b43b67bbfa844a9b015957c...@mmoexchmbs01.jhacorp.com Content-Type: text/plain; charset=us-ascii Hi Prabahar. It is still not working for me and I can't seem to find a working solution. Has anyone been able to get this to work? James Key From: Prabahar M [prabaha...@gmail.com] Sent: Tuesday, April 21, 2009 5:50 PM To: James Key Subject: Re: AAR and Offnet Transfers Hi James, Does it work for you? I tried and it is not working for me. If it works for you, could you send me the config steps please. Thansk, Prabahar Thanks for the reply Cliff. That is what I was thinking and did configure it this way, but when site B tries to transfer i get the message that transfer is not allowed. I will take a closer look. James Key From: Cliff McGlamry [cliff at mcglamry.net]
[OSL | CCIE_Voice] CME Transfer
I'm trying to setup a quick transfer to voicemail option. I'd like to have a scenario as follows: Extension 5001 receives a call. A speed dial is configured on the second button of this ephone (*5002) The user on 5001 presses transfer and the speed dial button for *5002. Pressing Transfer again is NOT needed. The call should go directly to voicemail greeting for 5002. I have the whole vmdirect thing setup with translation patterns so if I simply press the speed-dial on 5001, the call goes directly to 5002's greeting. I can't get a call from PSTN to be directly transferred to 5002's Voicemail greeting though. The PSTN call simply gets disconnected when I press the speed-dial button. I've tried a couple of options with call-forward and transfer pattern, but nothing works. Any ideas?
[OSL | CCIE_Voice] Digging for VATS
Can someone tell me where to look for the VATS configuration info ?I know I can do a search and find it but in case I have to go digging for it on lab day, I need to know how to navigate to it.
[OSL | CCIE_Voice] Call-forward vs transfer-pattern in UCME
Can someone help me understand what I'm missing regarding call-forward pattern and transfer-pattern in CME. I'd like to control where a call can be transferred to. Either as a consult transfer or using CfwdAll .
[OSL | CCIE_Voice] IPCC Prompt Variable Format.
Probably a very elementary question but it's never been clear to me. Regarding how prompt files are referenced within an IPCC script: How do it related to WHERE the prompt files are expected to be? Example QueuePrompt = SP[ICD\ICDQUEUE.wav] vs QueuePrompt = P[myprompt.wav]
[OSL | CCIE_Voice] Extension Mobility Question
Can someone point me towards info on how to approach EM to allow a user to move between HQ and BR1 but make sure 911 calls go out only the local GW.
Re: [OSL | CCIE_Voice] Extension Mobility Question
Alex, Thanks for the reply. I got it working. On Thu, Mar 26, 2009 at 2:46 PM, Alex alex.arsen...@gmail.com wrote: Have 911/9911 route patterns only in device CSS? Line CSS should not have any such pattern. Rgds Alex - Original Message - *From:* ccielab...@gmail.com *To:* ccie_voice@onlinestudylist.com *Sent:* Thursday, March 26, 2009 6:01 PM *Subject:* [OSL | CCIE_Voice] Extension Mobility Question Can someone point me towards info on how to approach EM to allow a user to move between HQ and BR1 but make sure 911 calls go out only the local GW.
[OSL | CCIE_Voice] Call forward to CUE failing.
I think I'm running into a bug but wanted to check if I might be missing something. I have a call coming into CME via the HQ-rtr GK. The call goes through fine if answered on the CME phone. (g729). If the call go noan to the CUE, I get a fast busy on the HQ phone. I also get a traceback msg on BR2. I have the voice services voip configured to allow H323 to SIP connections (all four options). Is there something else I'm missing?
[OSL | CCIE_Voice] Understanding Transcoding
I'm a little confused about how transcoding is invoked. Assuming I have an endpoint that makes a call (g.729) to another endpoint requiring g.711u. Is it ALWAYS the g.711 side that must allocate the transcoder resource?
Re: [OSL | CCIE_Voice] Gatekeeper blues :|
Yep, everything started working once I checked off requires MTP on the trunk. . But why? The call is coming in as g.729 and the device pool the trunk is in is set to g.729 On Mon, Mar 23, 2009 at 11:54 AM, Chris Parker cpar...@cparker.us wrote: If you are getting ring through when you place the call, then your GK config is probably OK. If its ringing that tells me the GK sent the call to the UCM. The fact that it goes fast busy when you pick up sounds like a codec/media problem to me. Check the media resources config and your regions. Chris ccielab...@gmail.com wrote: I'm working through scenario , trying to get 4 digit dial from BR2 to HQ working via the HQ GK. The GK config looks like this. gatekeeper zone local UCM ipexpert.com http://ipexpert.com 142.1.1.1 zone prefix UCM 1... gw-priority 10 Trunk_2 zone prefix UCM 1... gw-priority 9 Trunk_1 zone prefix UCM 1... gw-priority 0 cmegw zone prefix UCM 3... gw-priority 10 cmegw zone prefix UCM 3... gw-priority 0 Trunk_2 Trunk_1 gw-type-prefix 1#* gw ipaddr 142.2.64.12 1720 gw ipaddr 142.2.64.11 1720 no shutdown I have the UCM registering without a tech-prefix defined and I've adjusted the service parameter to insure the UCM listens on port 1720. The problem is the call rings through to the HQ DN but comes up fast busy when I answer the call. I've attempted to configure the Trunk from UCM as both an H.225 Trunk as well as an ICT trunk. I've unchecked Wait for far end h.245 terminal capability as well. Any help is greatly appreciated.
[OSL | CCIE_Voice] Local Directory on CME ?
Any trick to getting local directory to work on CME?I have Service local-directory configured and a handful of entries but can't get it to come up on the phone. I also have http server and http path configured. I've re-created the cnf files for good mesaure but nothing works.
[OSL | CCIE_Voice] Trunk Type.
Outside of being told specifically to configure a h.225 trunk vs. an IC Trunk, is there any reason to choose one over another in terms of connecting to the BR2 CME ?
Re: [OSL | CCIE_Voice] VMware ESXi for CCM servers
Any tricks to getting CUCM 7 working on ESXi ?My install keeps failing . I haven't invested too much time yet, as I'm trying hard to pass with the current blueprint :) On Sat, Mar 21, 2009 at 1:15 PM, Arun Kumar arunv...@gmail.com wrote: Hi I'm running CUCM 7 and Unity Connection 7 on ESXi with 6GB of RAM and 500GB of HDD and it's working fine. Not tested on Linux. Thanks On Sat, Mar 21, 2009 at 6:44 PM, WorkerBee cisco...@gmail.com wrote: Anyone has tried using ESXi with Quad core/8G ram instead of using VMware server on a Linux? Does ESXi gives a better performance? Thanks.
[OSL | CCIE_Voice] CME Gatekeeper Registration
Can someone confirm for me what the requirements are to have ephone-dn's in CME register to a gatekeeper. Is it dependent on using a dialplan command ? I keep getting inconsistent results. (I think) I also have run into the problem where I use No-Reg on the number configuration and it still ends up listed as an e.614 on the GK. Multiple reboots don't seem to clear up the problem either. This is driving me nuts.
[OSL | CCIE_Voice] Lab attack order.
I'm curious how people have approached the order of completing a lab. What order do you use to ensure gathering the most points and build the lab correctly without doubling back to adjust things as you get further into the lab. I know Vik and Mark have their own opinions on this, but I wanted to throw it out to the more general audience for feedback.
Re: [OSL | CCIE_Voice] Lab attack order.
Chris , Great layout and exactly what I was looking for. Thanks On Tue, Mar 3, 2009 at 10:27 PM, Chris Parker cpar...@cparker.us wrote: I think everyone has their own way of doing it, but here's how I go about it: STEP ZERO - read the whole lab carefully. It's hard to do when you are nervous. Try and pay attention the the details and look for tricks. Then work things out like what partitions and css you'll need, how the gatekeeper will work. Do you need multicast. Try and get you head around the lab as much as possible. 1. Gather info This is where I log into everything and look at CDP to get the MAC addresses of the phones and of the 6608 devices. I put all of this info into notepad for easy cut and paste. I hardly use any paper in the lab everything goes into notepad for easy access and no retyping. 2. 6500 Next I set up the 6500. I do all the vlans, aux vlans, voice ports, muulticast, and any QoS 3. HQ Here I set up interfaces, NTP, timezone, DHCP if need be, multicast if needed, QoS and the basic Gatekeeper config. I like doing all the QoS at the beginning of the lab, and by doing it on HQ first it helps with time because you can just cut and paste what you do on HQ to BR1 and BR2. I set up the GK enough so that I know the CME and UCM will register in the correct zones. Also if I see I need VIA-GK I go ahead and set up IPIPGW, dial peers and transcoder (as needed) on HQ. I go back and tweak the GK config later after I have set up UCM and CME. 4. BR1 I paste in QoS config copied from HQ, create the vlans, create interfaces, set helper address and multicast if needed, set up switch ports, set up dhcp if needed, set up ntp and timezone, set isdn switchtype, configure PRI for MGCP or H323, tweak isdn settings on serial, set up mgcp if needed, set up translation rules, set up CoR if needed, configure dial peers if needed for H323 and/or SRST, config transcoder and conf bridge if needed, set up SRST. 5. BR2 I paste in QoS config copied from HQ, create interfaces, set helper address and multicast if needed, set up dhcp if needed, set up ntp and timezone, set isdn switchtype, configure PRI, tweak isdn settings on serial, set voice service to allow h323 to sip, set up translation rules, set up CoR if needed, configure dial peers (these can often be copied with minor tweaks from BR1), config transcoder and conf bridge if needed (another section copied fro BR1), run telephony service setup, paste mac address from step 1 to appropriate ephone, set up H323 gateway and confirm registered to HQ, set voip dial peers for HQ GK and CUE, add ephone-dns for MWI, set up BACD, add any hunt groups as needed. 6. CUE I normally do this step while doing something else at the same time like setting up the 3550. There's alot of waiting in this step while CUE just does its thing. Once CUE is up and ready to be configured, I normally do the basic setup with the GUI (this will set up mwi, and your sip triggers and applications) and then do user creation from the CUE CLI. 7. 3550 Here I do vlans and set up ports, and do any QoS. 8. CME Testing So at this point 90% of the CLI work is done and all the CME/CUE stuff is complete. So here I do some quick testing on CME dialing in / out to PSTN, CUE MWI, and BACD. 9. Callmanager Basic So here Im doing all the CM stuff I can think of except call routing and phones. I go into serviceability first and turn on whatever services I need. Then I go back to admin. I use the Top Left to Right method. So System menu first and usually touch everything except Device Defaults. Then in Route Plan I do AAR, Partitions, CSS, and I try to do any translation patterns that I think I'll need. Then I jump over to the Device menu. I do any custom soft keys or button templates. I set up EM profiles if I need them. I set up all my gateways and make sure they register, I set up the GK and trunk I set up any CTI route points I need. I DO NOT set up the phones yet however. Then I move to the Feature Menu. I setup my phone services, voice mail, and park or pickup if need be. Then I go to the services menu. I setup all the media resources, lists and groups and make sure everything is registered and in the right device pool. After the media stuff is done I go back to the device pools and assign the MRGL. I always do IPMA or AC later after the phones are set up. 10. Eat Lunch Yep all the stuff listed above needs to be done before lunch. 11. Callmanager routing and Phones So this part is all about the Route Groups, Route Lists, and Route Patterns. I plan most of this out in step zero or at lunch. Once thats all done I start setting up the phones and users. If I did everything right in step 9 and I can do everything I need to on every phones without going back and add partitions or what not. When this is done I do some quick basic testing with the PSTN. 12. Gatekeeper Now that everything is built on CCM and CME
[OSL | CCIE_Voice] Parked Call problem
I'm working with call park. If I place an internal call (phone to phone) , I can park the call and pickup it up on another phone by dialing the call park DN. If I place a call from PSTN into HQ via the 6608 , I can park the call and pickup it up on another phone by dialing the call park DN. If I place a call from PSTN into HQ via the br1-rtr (MGCP) , I can park the call but get a message that your call can not be completed when I dial the call park DN. Any ideas?
[OSL | CCIE_Voice] Working on Unity Hunt group scenario.
I'm trying to get a Poor mans hunt group working by using Unity Call Handlers to ring a couple of extensions and then ultimately forward the call out to a PSTN phone.I've gotten everything working but the last forward out to a PSTN number (91408xxx) I've checked and changed the default restriction table in Unity , so I think the call should be allowed. Where can I look to troubleshoot this? The call viewer application doesn't show any good information. Thanks Scott
Re: [OSL | CCIE_Voice] Working on Unity Hunt group scenario.
Jose, The CSS was set correctly, but I never reset the ports :) It's all working now. BTW: On a semi-related note, What is the significance of the CSS assigned to the VM Pilot number? I can't see how that DN would ever be leveraged for outbound service. - Scott On Wed, Feb 18, 2009 at 10:46 AM, Jose Gregorio Linero (jlinero) jlin...@cisco.com wrote: Hi: Take a look at the CSS configuration in Unity ports. Regards, Jose -- *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of * ccielab...@gmail.com *Sent:* Miércoles, Febrero 18, 2009 10:45 AM *To:* OSL Group *Subject:* [OSL | CCIE_Voice] Working on Unity Hunt group scenario. I'm trying to get a Poor mans hunt group working by using Unity Call Handlers to ring a couple of extensions and then ultimately forward the call out to a PSTN phone. I've gotten everything working but the last forward out to a PSTN number (91408xxx) I've checked and changed the default restriction table in Unity , so I think the call should be allowed. Where can I look to troubleshoot this? The call viewer application doesn't show any good information. Thanks Scott
Re: [OSL | CCIE_Voice] Working on Unity Hunt group scenario.
Agreed, the CSS that is assigned to the VM port is used to determine what dial patterns are available. My question is the CSS that is assigned to the actual VM PILOT DN. As I see it, the PILOT DN wouldn't be used for outbound calling. - Scott On Wed, Feb 18, 2009 at 11:55 AM, Cliff McGlamry cl...@mcglamry.net wrote: You're making an outbound call from the port when you transfer to that last number. That uses the CSS to figure out how to route the call. - Original Message - *From:* ccielab...@gmail.com *To:* Jose Gregorio Linero (jlinero) jlin...@cisco.com *Cc:* OSL Group ccie_voice@onlinestudylist.com *Sent:* Wednesday, February 18, 2009 10:54 AM *Subject:* Re: [OSL | CCIE_Voice] Working on Unity Hunt group scenario. Jose, The CSS was set correctly, but I never reset the ports :) It's all working now. BTW: On a semi-related note, What is the significance of the CSS assigned to the VM Pilot number? I can't see how that DN would ever be leveraged for outbound service. - Scott On Wed, Feb 18, 2009 at 10:46 AM, Jose Gregorio Linero (jlinero) jlin...@cisco.com wrote: Hi: Take a look at the CSS configuration in Unity ports. Regards, Jose -- *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of * ccielab...@gmail.com *Sent:* Miércoles, Febrero 18, 2009 10:45 AM *To:* OSL Group *Subject:* [OSL | CCIE_Voice] Working on Unity Hunt group scenario. I'm trying to get a Poor mans hunt group working by using Unity Call Handlers to ring a couple of extensions and then ultimately forward the call out to a PSTN phone. I've gotten everything working but the last forward out to a PSTN number (91408xxx) I've checked and changed the default restriction table in Unity , so I think the call should be allowed. Where can I look to troubleshoot this? The call viewer application doesn't show any good information. Thanks Scott
Re: [OSL | CCIE_Voice] Antw: Re: Working on Unity Hunt group scenario.
That would make sense but it doesn't seem to work that way. I changed the CSS assigned to the VM PILOT to a CSS without any access to the internal partitions. All the VM ports were in an internal partition. When I dial the VM pilot, it shouldn't work based on my changes, but it does. - Scott On Wed, Feb 18, 2009 at 1:36 PM, Robert Schuknecht rschukne...@gmx.dewrote: Scott, as far as i know, the CSS of the VM-Pilot is used to reach the Hunt-Pilot of the Hunt-List/Line-Group of the VM-Ports. See the Voicemail-Pilot as a Speeddial to the Hunt-Pilot. /Robert ccielab...@gmail.com schrieb am Mittwoch, 18. Februar 2009 um 17:59 in Nachricht 884fa9c92a4d8b0e56bd24f2851093dc: Agreed, the CSS that is assigned to the VM port is used to determine what dial patterns are available. My question is the CSS that is assigned to the actual VM PILOT DN. As I see it, the PILOT DN wouldn't be used for outbound calling. - Scott On Wed, Feb 18, 2009 at 11:55 AM, Cliff McGlamry cl...@mcglamry.net wrote: You're making an outbound call from the port when you transfer to that last number. That uses the CSS to figure out how to route the call. - Original Message - *From:* ccielab...@gmail.com *To:* Jose Gregorio Linero (jlinero) jlin...@cisco.com *Cc:* OSL Group ccie_voice@onlinestudylist.com *Sent:* Wednesday, February 18, 2009 10:54 AM *Subject:* Re: [OSL | CCIE_Voice] Working on Unity Hunt group scenario. Jose, The CSS was set correctly, but I never reset the ports :) It's all working now. BTW: On a semi-related note, What is the significance of the CSS assigned to the VM Pilot number? I can't see how that DN would ever be leveraged for outbound service. - Scott On Wed, Feb 18, 2009 at 10:46 AM, Jose Gregorio Linero (jlinero) jlin...@cisco.com wrote: Hi: Take a look at the CSS configuration in Unity ports. Regards, Jose -- *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of * ccielab...@gmail.com *Sent:* Miércoles, Febrero 18, 2009 10:45 AM *To:* OSL Group *Subject:* [OSL | CCIE_Voice] Working on Unity Hunt group scenario. I'm trying to get a Poor mans hunt group working by using Unity Call Handlers to ring a couple of extensions and then ultimately forward the call out to a PSTN phone. I've gotten everything working but the last forward out to a PSTN number (91408xxx) I've checked and changed the default restriction table in Unity , so I think the call should be allowed. Where can I look to troubleshoot this? The call viewer application doesn't show any good information. Thanks Scott
[OSL | CCIE_Voice] Device Pool Question
I had been struggling with getting the IPIPGW scenario working (Task 4.9) in the IPEXPERT workbook. The specific problem was making an H.323 call from CME and have it delivered via SIP Trunk to CM. I found , although I had an MRGL assigned to the trunk, the MTP within the MRGL wasn't in the same Device Pool as the trunk. Once I assigned the MTP to the same Device Pool as the Trunk (DP_711only) , the call completed. This doesn't make sense to me. The MTP was in a device pool called Default which contained a region called default The Trunk was in a device pool called DP_711only which contained a region called g711 The regions were configured to use G.711 between each other. What am I not understanding here? What would require the MTP to be in the same Device pool? - Scott
Re: [OSL | CCIE_Voice] Device Pool Question
Thanks for the reply Mark.I'll give it a try. On Fri, Jan 9, 2009 at 1:16 PM, Mark Snow ms...@ipexpert.com wrote: Well with all as you said, it 'should' have worked fine. That being said, maybe at one time there was another DP assiged or a different R within one of the DPs. Sometimes the UCM 4 DB had to be 'bumped' to get it to reconize some changes and possibly your change did just that. If all is as you say then I would imagine that changing it back would also result in a completed call. HTH, Mark Snow Sr Technical Instructor IPexpert, Inc. Sent from my iPhone On Jan 9, 2009, at 11:52 AM, ccielab...@gmail.com wrote: I had been struggling with getting the IPIPGW scenario working (Task 4.9) in the IPEXPERT workbook. The specific problem was making an H.323 call from CME and have it delivered via SIP Trunk to CM. I found , although I had an MRGL assigned to the trunk, the MTP within the MRGL wasn't in the same Device Pool as the trunk. Once I assigned the MTP to the same Device Pool as the Trunk (DP_711only) , the call completed. This doesn't make sense to me. The MTP was in a device pool called Default which contained a region called default The Trunk was in a device pool called DP_711only which contained a region called g711 The regions were configured to use G.711 between each other. What am I not understanding here? What would require the MTP to be in the same Device pool? - Scott
[OSL | CCIE_Voice] Transcoding Question.
I need to get a better understanding of how transcoding is invoked. I've setup an IPIPGW on the HQ Router. I'm trying to setup H.323(g711) - SIP (g729) calls to/from Hq/SiteC I've configured a transcoder and registered it to the telephony service on the HQ RT. I've noticed in the IPExpert proctor guide that the HQ router is registered as a HW transcoder in the UCM. Is that needed? I assumed the a g711 from UCM would come into the IPIPGW and the IPIPGW would invoke a transcoder. I didn't think the UCM needed anything for this. Can someone help me out? - Scott
[OSL | CCIE_Voice] Correction: Transcoding Question.
After some review, I see that the MTP being configured is a software MTP . It is being assigned to the device pool that will be used for the ICT trunk going to the IPIPGw. -- Forwarded message -- From: ccielab...@gmail.com Date: Sat, Jan 3, 2009 at 11:17 AM Subject: Transcoding Question. To: OSL Group ccie_voice@onlinestudylist.com I need to get a better understanding of how transcoding is invoked. I've setup an IPIPGW on the HQ Router. I'm trying to setup H.323(g711) - SIP (g729) calls to/from Hq/SiteC I've configured a transcoder and registered it to the telephony service on the HQ RT. I've noticed in the IPExpert proctor guide that the HQ router is registered as a HW transcoder in the UCM. Is that needed? I assumed the a g711 from UCM would come into the IPIPGW and the IPIPGW would invoke a transcoder. I didn't think the UCM needed anything for this. Can someone help me out? - Scott
