[OSL | CCIE_Voice] Single Number Reach
I'm having a hard time when an internal extension calls another internal extension that uses SNR, the From phone number on the PSTN phone is 4 digits instead of 7. For example, extension 2001 calls 2003, and 2003 simultaneously rings a PSTN phone number. The display on the PSTN phone says HqPh1 (2001) instead of the 7 digit or 10 digit number. I have created PT_SNR which is assigned to CSS_SNR. I have CSS_SNR assigned to the Remote Destination Profile for both CSS and Redirecting CSS. My SNR number is +14086347694 and I have a route pattern that contains \+1408.6347694 which egresses the RL_HQ_ONLY (this is not Standard Local Route Group). I also created a Translation Pattern with PT_SNR and I have checked Use External Phone Number Mask. I was expecting this to take the 4 digit Calling number and insert the External mask instead. I tried following the steps in the Mock Lab guide (I believe it is Lab 6) but I still cannot get it working. Any assistance would be appreciated. Perhaps someone has a blog post? Thanks, Mark ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCX preventing from answering calls that it originated
Hi Miron, Thanks for this. There is even more: Enable for Supervised Transfers - Check this check box so that Cisco Unity Connection uses DTMF to detect and reject calls that have been transferred to another extension (by using supervised transfer) and that have been transferred back to Connection. If the call loop is not detected and rejected, Connection records a voice message that contains the prompt to leave a voice message. Default setting: Check box not checked. Enable for Forwarded Message Notification Calls (by Using Extension) Check this check box so that Cisco Unity Connection uses the extension to detect and reject new-message notifications that are sent to a device (such as a mobile phone) and that the device forwards back to Connection because the device did not answer. If the call loop is not detected and rejected, the call creates a new voice message for the user and triggers Connection to send a new-message notification call to the device. Default setting: Check box not checked. But these settings appear available on a UC Version 7.1.5ES7.1-7 not sure at the moment if this is also at 7.0 available, but I will know at the next proctorlabs session. ;-) Cheers, Bernhard Von: Miron Kobelski [mailto:findko...@gmail.com] Gesendet: Freitag, 1. Oktober 2010 00:13 An: Stutz, Bernhard Cc: ccie_voice@onlinestudylist.com Betreff: Re: [OSL | CCIE_Voice] UCX preventing from answering calls that it originated Hi Bernhard, My understanding is that CUC can sometimes originate calls, which could be routed back to it. E.g. voicemail notification for a DN with cfwdall back to CUC would end up as an endless loop of CUC speaking with itself. Have a look at options at: CUC Administration Telephony Integration Phone System Call Loop Detection by Using DTMF regards kobel On Thu, Sep 30, 2010 at 18:42, Stutz, Bernhard st...@pandacom.de wrote: Hi, I came across a question that is always within a Unity Connection integration: The UC server should be prevented from answering calls that it originated. I am not sure what this means. Could someone please shed some light on it? Is there a checkbox somewhere that needs to be checked in order to achieve this? I can't find a explanation for this at the solutions so far, so i was guessing that this is given anyway. thanks in advance, Bernhard ___ 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] Single Number Reach
Hi Mark, The EPNM does not work at RP for SNR. Have you try to set EPNM at RL level? TN. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] How to install CUCM
http://chikkis.blogspot.com/2009/10/installing-cucm-7-pub-and-sub-on-vmware.html This link will help you get thru. you can also go to you tube there are lots of videos on you tube, that will help you complete your installation successfully. Pithog oil ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Single Number Reach
If I'm following your example correctly, Mark, then you aren't hitting on the translation pattern. The SNR call is matching the \+1408.6347694 RP, to go out, why would it be hitting the translation pattern? Perhaps you meant to configure this as a Calling Party Transformation? mike On Fri, Oct 1, 2010 at 2:38 AM, Mark Holloway m...@markholloway.com wrote: I'm having a hard time when an internal extension calls another internal extension that uses SNR, the From phone number on the PSTN phone is 4 digits instead of 7. For example, extension 2001 calls 2003, and 2003 simultaneously rings a PSTN phone number. The display on the PSTN phone says HqPh1 (2001) instead of the 7 digit or 10 digit number. I have created PT_SNR which is assigned to CSS_SNR. I have CSS_SNR assigned to the Remote Destination Profile for both CSS and Redirecting CSS. My SNR number is +14086347694 and I have a route pattern that contains \+1408.6347694 which egresses the RL_HQ_ONLY (this is not Standard Local Route Group). I also created a Translation Pattern with PT_SNR and I have checked Use External Phone Number Mask. I was expecting this to take the 4 digit Calling number and insert the External mask instead. I tried following the steps in the Mock Lab guide (I believe it is Lab 6) but I still cannot get it working. Any assistance would be appreciated. Perhaps someone has a blog post? Thanks, Mark ___ 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] Single Number Reach
Just hit the same problem in Vol2 Lab4 and I can confirm that this doesn't work at the RP level but does work at the RL level. Is this a known bug ? Graham On 1 Oct 2010, at 13:35, Tam Nhu wrote: Hi Mark, The EPNM does not work at RP for SNR. Have you try to set EPNM at RL level? TN. ___ 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] Single Number Reach
Graham, same thing here. This is a summary of what I've done to get it working correctly. I eliminated using Translation Profiles as I didn't find them necessary for this. Create PT_SNR which is assigned to CSS_SNR Create a Remote Destination Profile and assign CSS_SNR to both Calling Search Space and Rerouting Calling Search Space. Build/associate your end user with this Remote Destination Profile. Build a Route List (RL_SNR) that includes just the HQ gateway and set the Calling Party External Phone Mask to On. Doing this in the Route Pattern won't work. Set Called Party to Subscriber (assuming the Remote Destination number is a local number). Lastly, build a Route Pattern that matches your Remote Destination Profile external number and assign it to PT_SNR and RL_SNR. The only thing about this method is that when calls from 2001 ring 2003 which rings the PSTN, this method is using the external mask which means HQ1's external mask is E164. Typically when a Subscriber call egresses the HQ gateway you would want the From number to be 7 digits. Are you guys putting a Calling Party Transformation on your HQ gateway to strip off the HQ area code for Subscriber calls? For all other purposes of presenting 7, 10, or E164, I have always used the Calling Party Transform in either the Route Pattern or Route List's Route Group. Thanks, Mark On Oct 1, 2010, at 7:25 AM, Graham Hopkins wrote: Just hit the same problem in Vol2 Lab4 and I can confirm that this doesn't work at the RP level but does work at the RL level. Is this a known bug ? Graham On 1 Oct 2010, at 13:35, Tam Nhu wrote: Hi Mark, The EPNM does not work at RP for SNR. Have you try to set EPNM at RL level? TN. ___ 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] Single Number Reach
Sorry, I meant Translation Patterns, not Profiles. Still working on the From number presentation. I'm assuming that if HQ1 calls HQ3 the PSTN phone should show a 7 digit From number, but if BR1 calls 2003 the PSTN should show a 10 digit From number. Would you guys agree? On Oct 1, 2010, at 8:56 AM, Mark Holloway wrote: Graham, same thing here. This is a summary of what I've done to get it working correctly. I eliminated using Translation Profiles as I didn't find them necessary for this. Create PT_SNR which is assigned to CSS_SNR Create a Remote Destination Profile and assign CSS_SNR to both Calling Search Space and Rerouting Calling Search Space. Build/associate your end user with this Remote Destination Profile. Build a Route List (RL_SNR) that includes just the HQ gateway and set the Calling Party External Phone Mask to On. Doing this in the Route Pattern won't work. Set Called Party to Subscriber (assuming the Remote Destination number is a local number). Lastly, build a Route Pattern that matches your Remote Destination Profile external number and assign it to PT_SNR and RL_SNR. The only thing about this method is that when calls from 2001 ring 2003 which rings the PSTN, this method is using the external mask which means HQ1's external mask is E164. Typically when a Subscriber call egresses the HQ gateway you would want the From number to be 7 digits. Are you guys putting a Calling Party Transformation on your HQ gateway to strip off the HQ area code for Subscriber calls? For all other purposes of presenting 7, 10, or E164, I have always used the Calling Party Transform in either the Route Pattern or Route List's Route Group. Thanks, Mark On Oct 1, 2010, at 7:25 AM, Graham Hopkins wrote: Just hit the same problem in Vol2 Lab4 and I can confirm that this doesn't work at the RP level but does work at the RL level. Is this a known bug ? Graham On 1 Oct 2010, at 13:35, Tam Nhu wrote: Hi Mark, The EPNM does not work at RP for SNR. Have you try to set EPNM at RL level? TN. ___ 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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CUPC client
hi all, is anyone know proctor rack has CUPC client installed in the PC ? tks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Single Number Reach
Well I'm just showing the full E.164 as that's what the lab I'm looking at looks for. However I guess you could strip the HQ area code at the gateway with the calling party transformation. In the real world (plan to visit that soon) then the remote destination is likely to be a mobile phone which isn't really local to any gateway - at least not here in the UK so would be a national call from anywhere. Graham On 1 Oct 2010, at 17:10, Mark Holloway wrote: Sorry, I meant Translation Patterns, not Profiles. Still working on the From number presentation. I'm assuming that if HQ1 calls HQ3 the PSTN phone should show a 7 digit From number, but if BR1 calls 2003 the PSTN should show a 10 digit From number. Would you guys agree? On Oct 1, 2010, at 8:56 AM, Mark Holloway wrote: Graham, same thing here. This is a summary of what I've done to get it working correctly. I eliminated using Translation Profiles as I didn't find them necessary for this. Create PT_SNR which is assigned to CSS_SNR Create a Remote Destination Profile and assign CSS_SNR to both Calling Search Space and Rerouting Calling Search Space. Build/associate your end user with this Remote Destination Profile. Build a Route List (RL_SNR) that includes just the HQ gateway and set the Calling Party External Phone Mask to On. Doing this in the Route Pattern won't work. Set Called Party to Subscriber (assuming the Remote Destination number is a local number). Lastly, build a Route Pattern that matches your Remote Destination Profile external number and assign it to PT_SNR and RL_SNR. The only thing about this method is that when calls from 2001 ring 2003 which rings the PSTN, this method is using the external mask which means HQ1's external mask is E164. Typically when a Subscriber call egresses the HQ gateway you would want the From number to be 7 digits. Are you guys putting a Calling Party Transformation on your HQ gateway to strip off the HQ area code for Subscriber calls? For all other purposes of presenting 7, 10, or E164, I have always used the Calling Party Transform in either the Route Pattern or Route List's Route Group. Thanks, Mark On Oct 1, 2010, at 7:25 AM, Graham Hopkins wrote: Just hit the same problem in Vol2 Lab4 and I can confirm that this doesn't work at the RP level but does work at the RL level. Is this a known bug ? Graham On 1 Oct 2010, at 13:35, Tam Nhu wrote: Hi Mark, The EPNM does not work at RP for SNR. Have you try to set EPNM at RL level? TN. ___ 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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Single Number Reach
The crazy thing is I tried this but I couldn't get it to work. PT_HQ_CALLING_OUT assigned to CSS_HQ_CALLING_OUT assigned to CALLING Number Transform on the Outbound portion of the HQ gateway. Calling Party Transformation: PT_HQ_CALLING_OUT pattern is \+1480.!(replace 480 with what your HQ area code is) Strip Predot That should make the outbound From number +14805552001 appear as 5552001 on the PSTN phone. and I should see 5552001 in the isdn q931 debug output. I'm still seeing the full E164 number. On Oct 1, 2010, at 9:22 AM, Graham Hopkins wrote: Well I'm just showing the full E.164 as that's what the lab I'm looking at looks for. However I guess you could strip the HQ area code at the gateway with the calling party transformation. In the real world (plan to visit that soon) then the remote destination is likely to be a mobile phone which isn't really local to any gateway - at least not here in the UK so would be a national call from anywhere. Graham On 1 Oct 2010, at 17:10, Mark Holloway wrote: Sorry, I meant Translation Patterns, not Profiles. Still working on the From number presentation. I'm assuming that if HQ1 calls HQ3 the PSTN phone should show a 7 digit From number, but if BR1 calls 2003 the PSTN should show a 10 digit From number. Would you guys agree? On Oct 1, 2010, at 8:56 AM, Mark Holloway wrote: Graham, same thing here. This is a summary of what I've done to get it working correctly. I eliminated using Translation Profiles as I didn't find them necessary for this. Create PT_SNR which is assigned to CSS_SNR Create a Remote Destination Profile and assign CSS_SNR to both Calling Search Space and Rerouting Calling Search Space. Build/associate your end user with this Remote Destination Profile. Build a Route List (RL_SNR) that includes just the HQ gateway and set the Calling Party External Phone Mask to On. Doing this in the Route Pattern won't work. Set Called Party to Subscriber (assuming the Remote Destination number is a local number). Lastly, build a Route Pattern that matches your Remote Destination Profile external number and assign it to PT_SNR and RL_SNR. The only thing about this method is that when calls from 2001 ring 2003 which rings the PSTN, this method is using the external mask which means HQ1's external mask is E164. Typically when a Subscriber call egresses the HQ gateway you would want the From number to be 7 digits. Are you guys putting a Calling Party Transformation on your HQ gateway to strip off the HQ area code for Subscriber calls? For all other purposes of presenting 7, 10, or E164, I have always used the Calling Party Transform in either the Route Pattern or Route List's Route Group. Thanks, Mark On Oct 1, 2010, at 7:25 AM, Graham Hopkins wrote: Just hit the same problem in Vol2 Lab4 and I can confirm that this doesn't work at the RP level but does work at the RL level. Is this a known bug ? Graham On 1 Oct 2010, at 13:35, Tam Nhu wrote: Hi Mark, The EPNM does not work at RP for SNR. Have you try to set EPNM at RL level? TN. ___ 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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] QOS solution required.
well, that is why we have this group for - to help each other and test our config and knowledge before the real test. so gang, for WAN, when you need MLP, just go to subinterface, enter bandwidth and go to DLCI and type: auto qos voip [trust] fr-atm (trust is optional, if nothing else is asked for, but if in example, marking and remarking is asked for then don't use trust at the end and you will get access lists as well and additional remarking class - all of that you can tweak later). in example: Router(config)#interface serial 0/0/0.100 Router(config)#bandwidth 384 Router(config-subif)# frame-relay interface-dlci 100 Router(config-fr-dlci)#auto qos voip trust fr-atm that will auto-create interface Virtual-Template100 and attach it properly to the interface. also will apply LFI and fragmentation for 10ms delay. just be sure to enter bandwidth before as shown above. after that you can go and tweak whatever you need. in example, mostly you will tweak your FR class to reflect 95% of CIR as it got created now with full speed. map-class frame-relay whatever it is named with autoqos frame-relay cir 364800 frame-relay bc 3648 frame-relay be 0 frame-relay mincir 364800 other approach, if you DON'T need MLP, will be to set your bandwidth again and under DLCI interface type 'auto qos voip [trust]' only , with leaving fr-atm out of it. then you will get all again configured but without PPP/virtual-template interface and again you can tweak above. if you need to shape it then you will have to turn off frame-relay traffic-shaping under main serial interface and create one middle-man policy-map to shape it. in example: policy-map SHAPE class class-default shape average 364800 3648 0 service-policy output here attach one which is created by autoqos). then go under your map-class frame-relay and attach above SHAPE to it (first remove the one attached by autoqos as that one you just now did attach to SHAPE, so now attach SHAPE to map-class instead - nested approach) ensure to configure frame-relay fragment 480 under that map-class frame-relay as well for FRF.12. On Thu, Sep 30, 2010 at 8:32 PM, Ashar Siddiqui siddas...@gmail.com wrote: Help yourself .. it would then be worth being CCIE.. Ash voice-gang voice-gang wrote: Lab in 5 days if anyone can help that would be appreciated On Thu, Sep 30, 2010 at 4:12 PM, voice-gang voice-gang mgcptroubleshoot...@gmail.com mailto:mgcptroubleshoot...@gmail.com wrote: 8.1 Switch QoS 2) On port fa 1/0/13 which is connected to HQ Phone 1, guarantee 32k for incoming SCCP signaling traffic. Excess traffic should be marked to DSCP 8 and then transmitted. By default, IP Phones mark SCCP signaling traffic to CS3. (3 points) 8.2 Link fragmentation and Interleaving There is 384k frame-relay PVC between HQ and SiteB. Configure R1 and R2 to enable MLP, link fragmentation and interleaving on this circuit. (2 points) Regarding the WAN qos if i will configure the HQ to SB my SC link become 56 K So i can achieve this task by adding the class map of T1 link to SC but do we have to do this or not. Regards MGCP ___ 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] Single Number Reach
calling party transformation is done without prefix \ On Fri, Oct 1, 2010 at 7:00 PM, Mark Holloway m...@markholloway.com wrote: The crazy thing is I tried this but I couldn't get it to work. PT_HQ_CALLING_OUT assigned to CSS_HQ_CALLING_OUT assigned to CALLING Number Transform on the Outbound portion of the HQ gateway. Calling Party Transformation: PT_HQ_CALLING_OUT pattern is \+1480.! (replace 480 with what your HQ area code is) Strip Predot That should make the outbound From number +14805552001 appear as 5552001 on the PSTN phone. and I should see 5552001 in the isdn q931 debug output. I'm still seeing the full E164 number. On Oct 1, 2010, at 9:22 AM, Graham Hopkins wrote: Well I'm just showing the full E.164 as that's what the lab I'm looking at looks for. However I guess you could strip the HQ area code at the gateway with the calling party transformation. In the real world (plan to visit that soon) then the remote destination is likely to be a mobile phone which isn't really local to any gateway - at least not here in the UK so would be a national call from anywhere. Graham On 1 Oct 2010, at 17:10, Mark Holloway wrote: Sorry, I meant Translation Patterns, not Profiles. Still working on the From number presentation. I'm assuming that if HQ1 calls HQ3 the PSTN phone should show a 7 digit From number, but if BR1 calls 2003 the PSTN should show a 10 digit From number. Would you guys agree? On Oct 1, 2010, at 8:56 AM, Mark Holloway wrote: Graham, same thing here. This is a summary of what I've done to get it working correctly. I eliminated using Translation Profiles as I didn't find them necessary for this. Create PT_SNR which is assigned to CSS_SNR Create a Remote Destination Profile and assign CSS_SNR to both Calling Search Space and Rerouting Calling Search Space. Build/associate your end user with this Remote Destination Profile. Build a Route List (RL_SNR) that includes just the HQ gateway and set the Calling Party External Phone Mask to On. Doing this in the Route Pattern won't work. Set Called Party to Subscriber (assuming the Remote Destination number is a local number). Lastly, build a Route Pattern that matches your Remote Destination Profile external number and assign it to PT_SNR and RL_SNR. The only thing about this method is that when calls from 2001 ring 2003 which rings the PSTN, this method is using the external mask which means HQ1's external mask is E164. Typically when a Subscriber call egresses the HQ gateway you would want the From number to be 7 digits. Are you guys putting a Calling Party Transformation on your HQ gateway to strip off the HQ area code for Subscriber calls? For all other purposes of presenting 7, 10, or E164, I have always used the Calling Party Transform in either the Route Pattern or Route List's Route Group. Thanks, Mark On Oct 1, 2010, at 7:25 AM, Graham Hopkins wrote: Just hit the same problem in Vol2 Lab4 and I can confirm that this doesn't work at the RP level but does work at the RL level. Is this a known bug ? Graham On 1 Oct 2010, at 13:35, Tam Nhu wrote: Hi Mark, The EPNM does not work at RP for SNR. Have you try to set EPNM at RL level? TN. ___ 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 ___ 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] Single Number Reach
When doing it under Call Routing Transformation Pattern Calling Party Transformation you have to use \+ When doing it on the Calling Party transform mask on a Route Pattern or Route List you don't use \ On Oct 1, 2010, at 10:44 AM, sisiaji wrote: calling party transformation is done without prefix \ On Fri, Oct 1, 2010 at 7:00 PM, Mark Holloway m...@markholloway.com wrote: The crazy thing is I tried this but I couldn't get it to work. PT_HQ_CALLING_OUT assigned to CSS_HQ_CALLING_OUT assigned to CALLING Number Transform on the Outbound portion of the HQ gateway. Calling Party Transformation: PT_HQ_CALLING_OUT pattern is \+1480.! (replace 480 with what your HQ area code is) Strip Predot That should make the outbound From number +14805552001 appear as 5552001 on the PSTN phone. and I should see 5552001 in the isdn q931 debug output. I'm still seeing the full E164 number. On Oct 1, 2010, at 9:22 AM, Graham Hopkins wrote: Well I'm just showing the full E.164 as that's what the lab I'm looking at looks for. However I guess you could strip the HQ area code at the gateway with the calling party transformation. In the real world (plan to visit that soon) then the remote destination is likely to be a mobile phone which isn't really local to any gateway - at least not here in the UK so would be a national call from anywhere. Graham On 1 Oct 2010, at 17:10, Mark Holloway wrote: Sorry, I meant Translation Patterns, not Profiles. Still working on the From number presentation. I'm assuming that if HQ1 calls HQ3 the PSTN phone should show a 7 digit From number, but if BR1 calls 2003 the PSTN should show a 10 digit From number. Would you guys agree? On Oct 1, 2010, at 8:56 AM, Mark Holloway wrote: Graham, same thing here. This is a summary of what I've done to get it working correctly. I eliminated using Translation Profiles as I didn't find them necessary for this. Create PT_SNR which is assigned to CSS_SNR Create a Remote Destination Profile and assign CSS_SNR to both Calling Search Space and Rerouting Calling Search Space. Build/associate your end user with this Remote Destination Profile. Build a Route List (RL_SNR) that includes just the HQ gateway and set the Calling Party External Phone Mask to On. Doing this in the Route Pattern won't work. Set Called Party to Subscriber (assuming the Remote Destination number is a local number). Lastly, build a Route Pattern that matches your Remote Destination Profile external number and assign it to PT_SNR and RL_SNR. The only thing about this method is that when calls from 2001 ring 2003 which rings the PSTN, this method is using the external mask which means HQ1's external mask is E164. Typically when a Subscriber call egresses the HQ gateway you would want the From number to be 7 digits. Are you guys putting a Calling Party Transformation on your HQ gateway to strip off the HQ area code for Subscriber calls? For all other purposes of presenting 7, 10, or E164, I have always used the Calling Party Transform in either the Route Pattern or Route List's Route Group. Thanks, Mark On Oct 1, 2010, at 7:25 AM, Graham Hopkins wrote: Just hit the same problem in Vol2 Lab4 and I can confirm that this doesn't work at the RP level but does work at the RL level. Is this a known bug ? Graham On 1 Oct 2010, at 13:35, Tam Nhu wrote: Hi Mark, The EPNM does not work at RP for SNR. Have you try to set EPNM at RL level? TN. ___ 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 ___ 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] QOS solution required.
Yeah it is but if you have tested some solution yourself already and need some guidance as to where you are making mistake. This list, I presume, is not for those who post questions with even exam points mentioned having no clue what they are asking about and then expecting a reply! Sorry I would not help! Ash sisiaji wrote: well, that is why we have this group for - to help each other and test our config and knowledge before the real test. so gang, for WAN, when you need MLP, just go to subinterface, enter bandwidth and go to DLCI and type: auto qos voip [trust] fr-atm (trust is optional, if nothing else is asked for, but if in example, marking and remarking is asked for then don't use trust at the end and you will get access lists as well and additional remarking class - all of that you can tweak later). in example: Router(config)#interface serial 0/0/0.100 Router(config)#bandwidth 384 Router(config-subif)# frame-relay interface-dlci 100 Router(config-fr-dlci)#auto qos voip trust fr-atm that will auto-create interface Virtual-Template100 and attach it properly to the interface. also will apply LFI and fragmentation for 10ms delay. just be sure to enter bandwidth before as shown above. after that you can go and tweak whatever you need. in example, mostly you will tweak your FR class to reflect 95% of CIR as it got created now with full speed. map-class frame-relay whatever it is named with autoqos frame-relay cir 364800 frame-relay bc 3648 frame-relay be 0 frame-relay mincir 364800 other approach, if you DON'T need MLP, will be to set your bandwidth again and under DLCI interface type 'auto qos voip [trust]' only , with leaving fr-atm out of it. then you will get all again configured but without PPP/virtual-template interface and again you can tweak above. if you need to shape it then you will have to turn off frame-relay traffic-shaping under main serial interface and create one middle-man policy-map to shape it. in example: policy-map SHAPE class class-default shape average 364800 3648 0 service-policy output here attach one which is created by autoqos). then go under your map-class frame-relay and attach above SHAPE to it (first remove the one attached by autoqos as that one you just now did attach to SHAPE, so now attach SHAPE to map-class instead - nested approach) ensure to configure frame-relay fragment 480 under that map-class frame-relay as well for FRF.12. On Thu, Sep 30, 2010 at 8:32 PM, Ashar Siddiqui siddas...@gmail.com mailto:siddas...@gmail.com wrote: Help yourself .. it would then be worth being CCIE.. Ash voice-gang voice-gang wrote: Lab in 5 days if anyone can help that would be appreciated On Thu, Sep 30, 2010 at 4:12 PM, voice-gang voice-gang mgcptroubleshoot...@gmail.com mailto:mgcptroubleshoot...@gmail.com mailto:mgcptroubleshoot...@gmail.com mailto:mgcptroubleshoot...@gmail.com wrote: 8.1 Switch QoS 2) On port fa 1/0/13 which is connected to HQ Phone 1, guarantee 32k for incoming SCCP signaling traffic. Excess traffic should be marked to DSCP 8 and then transmitted. By default, IP Phones mark SCCP signaling traffic to CS3. (3 points) 8.2 Link fragmentation and Interleaving There is 384k frame-relay PVC between HQ and SiteB. Configure R1 and R2 to enable MLP, link fragmentation and interleaving on this circuit. (2 points) Regarding the WAN qos if i will configure the HQ to SB my SC link become 56 K So i can achieve this task by adding the class map of T1 link to SC but do we have to do this or not. Regards MGCP ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] QOS solution required.
I feel that there has been too many people blatantly breaking their NDAs on this list lately...please ask questions related to your studies, not exact exam questions ! Then I know I will be glad to help and I'm sure there are more on the list who will too... -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ashar Siddiqui Sent: Friday, October 01, 2010 10:57 AM To: sisiaji Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] QOS solution required. Yeah it is but if you have tested some solution yourself already and need some guidance as to where you are making mistake. This list, I presume, is not for those who post questions with even exam points mentioned having no clue what they are asking about and then expecting a reply! Sorry I would not help! Ash sisiaji wrote: well, that is why we have this group for - to help each other and test our config and knowledge before the real test. so gang, for WAN, when you need MLP, just go to subinterface, enter bandwidth and go to DLCI and type: auto qos voip [trust] fr-atm (trust is optional, if nothing else is asked for, but if in example, marking and remarking is asked for then don't use trust at the end and you will get access lists as well and additional remarking class - all of that you can tweak later). in example: Router(config)#interface serial 0/0/0.100 Router(config)#bandwidth 384 Router(config-subif)# frame-relay interface-dlci 100 Router(config-fr-dlci)#auto qos voip trust fr-atm that will auto-create interface Virtual-Template100 and attach it properly to the interface. also will apply LFI and fragmentation for 10ms delay. just be sure to enter bandwidth before as shown above. after that you can go and tweak whatever you need. in example, mostly you will tweak your FR class to reflect 95% of CIR as it got created now with full speed. map-class frame-relay whatever it is named with autoqos frame-relay cir 364800 frame-relay bc 3648 frame-relay be 0 frame-relay mincir 364800 other approach, if you DON'T need MLP, will be to set your bandwidth again and under DLCI interface type 'auto qos voip [trust]' only , with leaving fr-atm out of it. then you will get all again configured but without PPP/virtual-template interface and again you can tweak above. if you need to shape it then you will have to turn off frame-relay traffic-shaping under main serial interface and create one middle-man policy-map to shape it. in example: policy-map SHAPE class class-default shape average 364800 3648 0 service-policy output here attach one which is created by autoqos). then go under your map-class frame-relay and attach above SHAPE to it (first remove the one attached by autoqos as that one you just now did attach to SHAPE, so now attach SHAPE to map-class instead - nested approach) ensure to configure frame-relay fragment 480 under that map-class frame-relay as well for FRF.12. On Thu, Sep 30, 2010 at 8:32 PM, Ashar Siddiqui siddas...@gmail.com mailto:siddas...@gmail.com wrote: Help yourself .. it would then be worth being CCIE.. Ash voice-gang voice-gang wrote: Lab in 5 days if anyone can help that would be appreciated On Thu, Sep 30, 2010 at 4:12 PM, voice-gang voice-gang mgcptroubleshoot...@gmail.com mailto:mgcptroubleshoot...@gmail.com mailto:mgcptroubleshoot...@gmail.com mailto:mgcptroubleshoot...@gmail.com wrote: 8.1 Switch QoS 2) On port fa 1/0/13 which is connected to HQ Phone 1, guarantee 32k for incoming SCCP signaling traffic. Excess traffic should be marked to DSCP 8 and then transmitted. By default, IP Phones mark SCCP signaling traffic to CS3. (3 points) 8.2 Link fragmentation and Interleaving There is 384k frame-relay PVC between HQ and SiteB. Configure R1 and R2 to enable MLP, link fragmentation and interleaving on this circuit. (2 points) Regarding the WAN qos if i will configure the HQ to SB my SC link become 56 K So i can achieve this task by adding the class map of T1 link to SC but do we have to do this or not. Regards MGCP ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com
Re: [OSL | CCIE_Voice] Single Number Reach
yeah, you are right, I was referring to RP/RL transformations... i tested it and i got the same in my lab so i guess, as you already mentioned before, the way to do it is to actually put Calling Party Transform Mask to be XXX on the RL (for RG member). On Fri, Oct 1, 2010 at 7:52 PM, Mark Holloway m...@markholloway.com wrote: When doing it under Call Routing Transformation Pattern Calling Party Transformation you have to use \+ When doing it on the Calling Party transform mask on a Route Pattern or Route List you don't use \ On Oct 1, 2010, at 10:44 AM, sisiaji wrote: calling party transformation is done without prefix \ On Fri, Oct 1, 2010 at 7:00 PM, Mark Holloway m...@markholloway.com wrote: The crazy thing is I tried this but I couldn't get it to work. PT_HQ_CALLING_OUT assigned to CSS_HQ_CALLING_OUT assigned to CALLING Number Transform on the Outbound portion of the HQ gateway. Calling Party Transformation: PT_HQ_CALLING_OUT pattern is \+1480.! (replace 480 with what your HQ area code is) Strip Predot That should make the outbound From number +14805552001 appear as 5552001 on the PSTN phone. and I should see 5552001 in the isdn q931 debug output. I'm still seeing the full E164 number. On Oct 1, 2010, at 9:22 AM, Graham Hopkins wrote: Well I'm just showing the full E.164 as that's what the lab I'm looking at looks for. However I guess you could strip the HQ area code at the gateway with the calling party transformation. In the real world (plan to visit that soon) then the remote destination is likely to be a mobile phone which isn't really local to any gateway - at least not here in the UK so would be a national call from anywhere. Graham On 1 Oct 2010, at 17:10, Mark Holloway wrote: Sorry, I meant Translation Patterns, not Profiles. Still working on the From number presentation. I'm assuming that if HQ1 calls HQ3 the PSTN phone should show a 7 digit From number, but if BR1 calls 2003 the PSTN should show a 10 digit From number. Would you guys agree? On Oct 1, 2010, at 8:56 AM, Mark Holloway wrote: Graham, same thing here. This is a summary of what I've done to get it working correctly. I eliminated using Translation Profiles as I didn't find them necessary for this. Create PT_SNR which is assigned to CSS_SNR Create a Remote Destination Profile and assign CSS_SNR to both Calling Search Space and Rerouting Calling Search Space. Build/associate your end user with this Remote Destination Profile. Build a Route List (RL_SNR) that includes just the HQ gateway and set the Calling Party External Phone Mask to On. Doing this in the Route Pattern won't work. Set Called Party to Subscriber (assuming the Remote Destination number is a local number). Lastly, build a Route Pattern that matches your Remote Destination Profile external number and assign it to PT_SNR and RL_SNR. The only thing about this method is that when calls from 2001 ring 2003 which rings the PSTN, this method is using the external mask which means HQ1's external mask is E164. Typically when a Subscriber call egresses the HQ gateway you would want the From number to be 7 digits. Are you guys putting a Calling Party Transformation on your HQ gateway to strip off the HQ area code for Subscriber calls? For all other purposes of presenting 7, 10, or E164, I have always used the Calling Party Transform in either the Route Pattern or Route List's Route Group. Thanks, Mark On Oct 1, 2010, at 7:25 AM, Graham Hopkins wrote: Just hit the same problem in Vol2 Lab4 and I can confirm that this doesn't work at the RP level but does work at the RL level. Is this a known bug ? Graham On 1 Oct 2010, at 13:35, Tam Nhu wrote: Hi Mark, The EPNM does not work at RP for SNR. Have you try to set EPNM at RL level? TN. ___ 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 ___ 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] Single Number Reach
The only issue with this is you don't know if the calling party is Subscriber, National, or International, so you can't use XXX because if BR2 or BR1 calls HQ3 the From number would only show the first 7 digits. On Oct 1, 2010, at 11:21 AM, sisiaji wrote: yeah, you are right, I was referring to RP/RL transformations... i tested it and i got the same in my lab so i guess, as you already mentioned before, the way to do it is to actually put Calling Party Transform Mask to be XXX on the RL (for RG member). On Fri, Oct 1, 2010 at 7:52 PM, Mark Holloway m...@markholloway.com wrote: When doing it under Call Routing Transformation Pattern Calling Party Transformation you have to use \+ When doing it on the Calling Party transform mask on a Route Pattern or Route List you don't use \ On Oct 1, 2010, at 10:44 AM, sisiaji wrote: calling party transformation is done without prefix \ On Fri, Oct 1, 2010 at 7:00 PM, Mark Holloway m...@markholloway.com wrote: The crazy thing is I tried this but I couldn't get it to work. PT_HQ_CALLING_OUT assigned to CSS_HQ_CALLING_OUT assigned to CALLING Number Transform on the Outbound portion of the HQ gateway. Calling Party Transformation: PT_HQ_CALLING_OUT pattern is \+1480.! (replace 480 with what your HQ area code is) Strip Predot That should make the outbound From number +14805552001 appear as 5552001 on the PSTN phone. and I should see 5552001 in the isdn q931 debug output. I'm still seeing the full E164 number. On Oct 1, 2010, at 9:22 AM, Graham Hopkins wrote: Well I'm just showing the full E.164 as that's what the lab I'm looking at looks for. However I guess you could strip the HQ area code at the gateway with the calling party transformation. In the real world (plan to visit that soon) then the remote destination is likely to be a mobile phone which isn't really local to any gateway - at least not here in the UK so would be a national call from anywhere. Graham On 1 Oct 2010, at 17:10, Mark Holloway wrote: Sorry, I meant Translation Patterns, not Profiles. Still working on the From number presentation. I'm assuming that if HQ1 calls HQ3 the PSTN phone should show a 7 digit From number, but if BR1 calls 2003 the PSTN should show a 10 digit From number. Would you guys agree? On Oct 1, 2010, at 8:56 AM, Mark Holloway wrote: Graham, same thing here. This is a summary of what I've done to get it working correctly. I eliminated using Translation Profiles as I didn't find them necessary for this. Create PT_SNR which is assigned to CSS_SNR Create a Remote Destination Profile and assign CSS_SNR to both Calling Search Space and Rerouting Calling Search Space. Build/associate your end user with this Remote Destination Profile. Build a Route List (RL_SNR) that includes just the HQ gateway and set the Calling Party External Phone Mask to On. Doing this in the Route Pattern won't work. Set Called Party to Subscriber (assuming the Remote Destination number is a local number). Lastly, build a Route Pattern that matches your Remote Destination Profile external number and assign it to PT_SNR and RL_SNR. The only thing about this method is that when calls from 2001 ring 2003 which rings the PSTN, this method is using the external mask which means HQ1's external mask is E164. Typically when a Subscriber call egresses the HQ gateway you would want the From number to be 7 digits. Are you guys putting a Calling Party Transformation on your HQ gateway to strip off the HQ area code for Subscriber calls? For all other purposes of presenting 7, 10, or E164, I have always used the Calling Party Transform in either the Route Pattern or Route List's Route Group. Thanks, Mark On Oct 1, 2010, at 7:25 AM, Graham Hopkins wrote: Just hit the same problem in Vol2 Lab4 and I can confirm that this doesn't work at the RP level but does work at the RL level. Is this a known bug ? Graham On 1 Oct 2010, at 13:35, Tam Nhu wrote: Hi Mark, The EPNM does not work at RP for SNR. Have you try to set EPNM at RL level? TN. ___ 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 ___ 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] QOS solution required.
are you sure those are real questions? i thought they are pasting some tasks from ipexpert or other's mock labs or so...as i myself don't have those mock labs... how can i distinguish between those? On Fri, Oct 1, 2010 at 8:15 PM, CCIE Voice GMAIL givemeccievoice2...@gmail.com wrote: I feel that there has been too many people blatantly breaking their NDAs on this list lately...please ask questions related to your studies, not exact exam questions ! Then I know I will be glad to help and I'm sure there are more on the list who will too... -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ashar Siddiqui Sent: Friday, October 01, 2010 10:57 AM To: sisiaji Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] QOS solution required. Yeah it is but if you have tested some solution yourself already and need some guidance as to where you are making mistake. This list, I presume, is not for those who post questions with even exam points mentioned having no clue what they are asking about and then expecting a reply! Sorry I would not help! Ash sisiaji wrote: well, that is why we have this group for - to help each other and test our config and knowledge before the real test. so gang, for WAN, when you need MLP, just go to subinterface, enter bandwidth and go to DLCI and type: auto qos voip [trust] fr-atm (trust is optional, if nothing else is asked for, but if in example, marking and remarking is asked for then don't use trust at the end and you will get access lists as well and additional remarking class - all of that you can tweak later). in example: Router(config)#interface serial 0/0/0.100 Router(config)#bandwidth 384 Router(config-subif)# frame-relay interface-dlci 100 Router(config-fr-dlci)#auto qos voip trust fr-atm that will auto-create interface Virtual-Template100 and attach it properly to the interface. also will apply LFI and fragmentation for 10ms delay. just be sure to enter bandwidth before as shown above. after that you can go and tweak whatever you need. in example, mostly you will tweak your FR class to reflect 95% of CIR as it got created now with full speed. map-class frame-relay whatever it is named with autoqos frame-relay cir 364800 frame-relay bc 3648 frame-relay be 0 frame-relay mincir 364800 other approach, if you DON'T need MLP, will be to set your bandwidth again and under DLCI interface type 'auto qos voip [trust]' only , with leaving fr-atm out of it. then you will get all again configured but without PPP/virtual-template interface and again you can tweak above. if you need to shape it then you will have to turn off frame-relay traffic-shaping under main serial interface and create one middle-man policy-map to shape it. in example: policy-map SHAPE class class-default shape average 364800 3648 0 service-policy output here attach one which is created by autoqos). then go under your map-class frame-relay and attach above SHAPE to it (first remove the one attached by autoqos as that one you just now did attach to SHAPE, so now attach SHAPE to map-class instead - nested approach) ensure to configure frame-relay fragment 480 under that map-class frame-relay as well for FRF.12. On Thu, Sep 30, 2010 at 8:32 PM, Ashar Siddiqui siddas...@gmail.com mailto:siddas...@gmail.com wrote: Help yourself .. it would then be worth being CCIE.. Ash voice-gang voice-gang wrote: Lab in 5 days if anyone can help that would be appreciated On Thu, Sep 30, 2010 at 4:12 PM, voice-gang voice-gang mgcptroubleshoot...@gmail.com mailto:mgcptroubleshoot...@gmail.com mailto:mgcptroubleshoot...@gmail.com mailto:mgcptroubleshoot...@gmail.com wrote: 8.1 Switch QoS 2) On port fa 1/0/13 which is connected to HQ Phone 1, guarantee 32k for incoming SCCP signaling traffic. Excess traffic should be marked to DSCP 8 and then transmitted. By default, IP Phones mark SCCP signaling traffic to CS3. (3 points) 8.2 Link fragmentation and Interleaving There is 384k frame-relay PVC between HQ and SiteB. Configure R1 and R2 to enable MLP, link fragmentation and interleaving on this circuit. (2 points) Regarding the WAN qos if i will configure the HQ to SB my SC link become 56 K So i can achieve this task by adding the class map of T1 link to SC but do we have to do this or not. Regards MGCP
Re: [OSL | CCIE_Voice] Single Number Reach
Same here , I was beginning to think that no patterns are matched in calling number transformations - but I tested with a pattern of ! and a mask of 12345 and that works. So it would appear that there is a mismatch between \+1480.! and the calling number, which does seem odd as if you leave it alone it gets sent to the PSTN as +1480XXX. It would appear that it should match as the pattern ! with XXX works, but as Mark says this doesn't do what he requires Graham On 1 Oct 2010, at 19:23, Mark Holloway wrote: The only issue with this is you don't know if the calling party is Subscriber, National, or International, so you can't use XXX because if BR2 or BR1 calls HQ3 the From number would only show the first 7 digits. On Oct 1, 2010, at 11:21 AM, sisiaji wrote: yeah, you are right, I was referring to RP/RL transformations... i tested it and i got the same in my lab so i guess, as you already mentioned before, the way to do it is to actually put Calling Party Transform Mask to be XXX on the RL (for RG member). On Fri, Oct 1, 2010 at 7:52 PM, Mark Holloway m...@markholloway.com wrote: When doing it under Call Routing Transformation Pattern Calling Party Transformation you have to use \+ When doing it on the Calling Party transform mask on a Route Pattern or Route List you don't use \ On Oct 1, 2010, at 10:44 AM, sisiaji wrote: calling party transformation is done without prefix \ On Fri, Oct 1, 2010 at 7:00 PM, Mark Holloway m...@markholloway.com wrote: The crazy thing is I tried this but I couldn't get it to work. PT_HQ_CALLING_OUT assigned to CSS_HQ_CALLING_OUT assigned to CALLING Number Transform on the Outbound portion of the HQ gateway. Calling Party Transformation: PT_HQ_CALLING_OUT pattern is \+1480.! (replace 480 with what your HQ area code is) Strip Predot That should make the outbound From number +14805552001 appear as 5552001 on the PSTN phone. and I should see 5552001 in the isdn q931 debug output. I'm still seeing the full E164 number. On Oct 1, 2010, at 9:22 AM, Graham Hopkins wrote: Well I'm just showing the full E.164 as that's what the lab I'm looking at looks for. However I guess you could strip the HQ area code at the gateway with the calling party transformation. In the real world (plan to visit that soon) then the remote destination is likely to be a mobile phone which isn't really local to any gateway - at least not here in the UK so would be a national call from anywhere. Graham On 1 Oct 2010, at 17:10, Mark Holloway wrote: Sorry, I meant Translation Patterns, not Profiles. Still working on the From number presentation. I'm assuming that if HQ1 calls HQ3 the PSTN phone should show a 7 digit From number, but if BR1 calls 2003 the PSTN should show a 10 digit From number. Would you guys agree? On Oct 1, 2010, at 8:56 AM, Mark Holloway wrote: Graham, same thing here. This is a summary of what I've done to get it working correctly. I eliminated using Translation Profiles as I didn't find them necessary for this. Create PT_SNR which is assigned to CSS_SNR Create a Remote Destination Profile and assign CSS_SNR to both Calling Search Space and Rerouting Calling Search Space. Build/associate your end user with this Remote Destination Profile. Build a Route List (RL_SNR) that includes just the HQ gateway and set the Calling Party External Phone Mask to On. Doing this in the Route Pattern won't work. Set Called Party to Subscriber (assuming the Remote Destination number is a local number). Lastly, build a Route Pattern that matches your Remote Destination Profile external number and assign it to PT_SNR and RL_SNR. The only thing about this method is that when calls from 2001 ring 2003 which rings the PSTN, this method is using the external mask which means HQ1's external mask is E164. Typically when a Subscriber call egresses the HQ gateway you would want the From number to be 7 digits. Are you guys putting a Calling Party Transformation on your HQ gateway to strip off the HQ area code for Subscriber calls? For all other purposes of presenting 7, 10, or E164, I have always used the Calling Party Transform in either the Route Pattern or Route List's Route Group. Thanks, Mark On Oct 1, 2010, at 7:25 AM, Graham Hopkins wrote: Just hit the same problem in Vol2 Lab4 and I can confirm that this doesn't work at the RP level but does work at the RL level. Is this a known bug ? Graham On 1 Oct 2010, at 13:35, Tam Nhu wrote: Hi Mark, The EPNM does not work at RP for SNR. Have you try to set EPNM at RL level? TN. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry
Re: [OSL | CCIE_Voice] QOS solution required.
I may be wrong in this particular instance, but if you've been paying attention to the list for the past 3 months or so you've seen it happen more than a few times. I apologize if this isn't the case here. However, I agree with Ash, this isn't a list for you to just get answers, this is a list to collaborate and solve problems. From: sisiaji [mailto:si.si.aj.i.v...@gmail.com] Sent: Friday, October 01, 2010 11:26 AM To: CCIE Voice GMAIL Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] QOS solution required. are you sure those are real questions? i thought they are pasting some tasks from ipexpert or other's mock labs or so...as i myself don't have those mock labs... how can i distinguish between those? On Fri, Oct 1, 2010 at 8:15 PM, CCIE Voice GMAIL givemeccievoice2...@gmail.com wrote: I feel that there has been too many people blatantly breaking their NDAs on this list lately...please ask questions related to your studies, not exact exam questions ! Then I know I will be glad to help and I'm sure there are more on the list who will too... -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ashar Siddiqui Sent: Friday, October 01, 2010 10:57 AM To: sisiaji Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] QOS solution required. Yeah it is but if you have tested some solution yourself already and need some guidance as to where you are making mistake. This list, I presume, is not for those who post questions with even exam points mentioned having no clue what they are asking about and then expecting a reply! Sorry I would not help! Ash sisiaji wrote: well, that is why we have this group for - to help each other and test our config and knowledge before the real test. so gang, for WAN, when you need MLP, just go to subinterface, enter bandwidth and go to DLCI and type: auto qos voip [trust] fr-atm (trust is optional, if nothing else is asked for, but if in example, marking and remarking is asked for then don't use trust at the end and you will get access lists as well and additional remarking class - all of that you can tweak later). in example: Router(config)#interface serial 0/0/0.100 Router(config)#bandwidth 384 Router(config-subif)# frame-relay interface-dlci 100 Router(config-fr-dlci)#auto qos voip trust fr-atm that will auto-create interface Virtual-Template100 and attach it properly to the interface. also will apply LFI and fragmentation for 10ms delay. just be sure to enter bandwidth before as shown above. after that you can go and tweak whatever you need. in example, mostly you will tweak your FR class to reflect 95% of CIR as it got created now with full speed. map-class frame-relay whatever it is named with autoqos frame-relay cir 364800 frame-relay bc 3648 frame-relay be 0 frame-relay mincir 364800 other approach, if you DON'T need MLP, will be to set your bandwidth again and under DLCI interface type 'auto qos voip [trust]' only , with leaving fr-atm out of it. then you will get all again configured but without PPP/virtual-template interface and again you can tweak above. if you need to shape it then you will have to turn off frame-relay traffic-shaping under main serial interface and create one middle-man policy-map to shape it. in example: policy-map SHAPE class class-default shape average 364800 3648 0 service-policy output here attach one which is created by autoqos). then go under your map-class frame-relay and attach above SHAPE to it (first remove the one attached by autoqos as that one you just now did attach to SHAPE, so now attach SHAPE to map-class instead - nested approach) ensure to configure frame-relay fragment 480 under that map-class frame-relay as well for FRF.12. On Thu, Sep 30, 2010 at 8:32 PM, Ashar Siddiqui siddas...@gmail.com mailto:siddas...@gmail.com wrote: Help yourself .. it would then be worth being CCIE.. Ash voice-gang voice-gang wrote: Lab in 5 days if anyone can help that would be appreciated On Thu, Sep 30, 2010 at 4:12 PM, voice-gang voice-gang mgcptroubleshoot...@gmail.com mailto:mgcptroubleshoot...@gmail.com mailto:mgcptroubleshoot...@gmail.com mailto:mgcptroubleshoot...@gmail.com wrote: 8.1 Switch QoS 2) On port fa 1/0/13 which is connected to HQ Phone 1, guarantee 32k for incoming SCCP signaling traffic. Excess traffic should be marked to DSCP 8 and then transmitted. By default, IP Phones mark SCCP signaling traffic to CS3. (3 points) 8.2 Link fragmentation and Interleaving There is 384k frame-relay PVC between HQ and SiteB. Configure R1 and R2 to enable MLP, link fragmentation and
Re: [OSL | CCIE_Voice] Speed for taking the Lab
Also highly recommended: Vik's Voice Lab Strategy vLecture, available via IPexpert's Facebook page: http://www.facebook.com/pages/IPexpert/24586557119?v=app_7146470109 Direct link: http://ipexpert.acrobat.com/p93148979/ cheers, sd From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Daniel Berlinski Sent: Thursday, September 30, 2010 5:23 PM To: Amp Cc: ccie_voice@onlinestudylist.com; Pithog Oil Subject: Re: [OSL | CCIE_Voice] Speed for taking the Lab HI Pithog Here comes my suggestion: Choose one lab and change the IP addresses in your environment, the number plan, and the physical position of the phones you work with on your desk, introduce infrastructure probs, make sure you make it very hard for your phones to register. Review the troubleshooting IP Tel book on chapter 3 I think speaks about phone registration and there is a white paper on cisco.com that talks about common phone registration probs, make sure you are familiar with those and practice those scenarios so you can see the symptoms happening before you in the stress room chamber On Fri, Oct 1, 2010 at 9:38 AM, Amp amccar...@cciequest.com wrote: Hey Pithog, You ask a tough question my friend. I think some of the things that you need to consider are how well do you know the core technologies and how fast can you correctly configure them? Based upon the forums and the practice labs there are going to be some things that you will need to know how to configure rather swiftly. Will you have CME with SCCP and SIP phones to configure on your lab? Who knows but it would be a good idea to know how to configure CME in a matter of minutes. Can you configure IOS media resources as fast as you can type your name? If not then ask yourself why not. Start configuring H323, MGCP, and Gatekeepers in notepad. If you can do it in notepad with little to no screw-ups then you can do it in the router lightning fast. Are you able to read the question and not over-complicate what's being asked? How fast can you configure COR? Furthermore what's your strategy? Do you plan on configuring once and copying, modifying, and pasting? What do you know really well and what do you need help in? Spend as much time trying to master the areas that you are weak in. Also remember, it is very possible that the IPX labs are more difficult than the actual lab so if you can't get through the IPX labs in less than 8 hours, can you do the core of what's being asked in a timely manner? So in my opinion, configure as much as you can in notepad to ensure you know the configuration steps inside and out. During your steps write out the steps to configure what's being asked. Do this without looking it up and see where you are coming up short. I know I didn't directly answer your question but I hope that helps. Amp Quoting Pithog Oil pithog...@yahoo.com: Please i will like to know if my speed is okay and good enough for the exam, it takes me 8 hours at the moment to finish the ipexpert, Labs, suggestions are welcome on how i can shorthen the time to 4 hours, i really hope its possible, please i need assistance on this. Ultimately i want to know how to manage my time better. 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] Single Number Reach
Remote Destination Profile has a Rerouting Calling Search Space. -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Holloway Sent: Friday, October 01, 2010 3:39 AM To: osl osl Subject: [OSL | CCIE_Voice] Single Number Reach I'm having a hard time when an internal extension calls another internal extension that uses SNR, the From phone number on the PSTN phone is 4 digits instead of 7. For example, extension 2001 calls 2003, and 2003 simultaneously rings a PSTN phone number. The display on the PSTN phone says HqPh1 (2001) instead of the 7 digit or 10 digit number. I have created PT_SNR which is assigned to CSS_SNR. I have CSS_SNR assigned to the Remote Destination Profile for both CSS and Redirecting CSS. My SNR number is +14086347694 and I have a route pattern that contains \+1408.6347694 which egresses the RL_HQ_ONLY (this is not Standard Local Route Group). I also created a Translation Pattern with PT_SNR and I have checked Use External Phone Number Mask. I was expecting this to take the 4 digit Calling number and insert the External mask instead. I tried following the steps in the Mock Lab guide (I believe it is Lab 6) but I still cannot get it working. Any assistance would be appreciated. Perhaps someone has a blog post? Thanks, Mark ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] +Dialing to Site C without Gatekeeper
Hello Experts, I'm working on TEHO to Site C with Gatekeeper. Everything works except for the + sign. On the debug ISDN Q931 on both sites, I see the + being sent and received, but on the phone it only shows my 11 digits (12123945003), but on the bottom of the phone it shows my plus sign. Any ideas??? Thanks in advance Cheers ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] +Dialing to Site C without Gatekeeper
That's a bug in CME I believe. That is the correct behavior. Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ayman labib Sent: Friday, October 01, 2010 2:37 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] +Dialing to Site C without Gatekeeper Hello Experts, I'm working on TEHO to Site C with Gatekeeper. Everything works except for the + sign. On the debug ISDN Q931 on both sites, I see the + being sent and received, but on the phone it only shows my 11 digits (12123945003), but on the bottom of the phone it shows my plus sign. Any ideas??? Thanks in advance CheersError! Filename not specified. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] SIP Phones in CME
Hi everyone, I am having a hard time remembering what command will affect the number displayed in the upper-right of the phones for CME. With SCCP, I know the description command will effect that number. How do you change this value for SIP phones registered to CME? Thanks for the help, Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP Phones in CME
description in voice register pool config mode. On Sat, Oct 2, 2010 at 10:51 AM, CCIE Voice GMAIL givemeccievoice2...@gmail.com wrote: Hi everyone, I am having a hard time remembering what command will affect the number displayed in the upper-right of the phones for CME. With SCCP, I know the description command will effect that number. How do you change this value for SIP phones registered to CME? Thanks for the help, Jeff ___ 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] SIP Phones in CME
Hi Dan, Thanks for the response. Before I could test what you had told me, I must have screwed something up. I keep getting a message on both of my CME phones saying Unprovisioned. I have reloaded my router and re-configured everything again but I am still getting that message. Has anyone seen this before? Jeff From: Daniel Berlinski [mailto:dberlin...@gmail.com] Sent: Friday, October 01, 2010 2:55 PM To: CCIE Voice GMAIL Cc: osl osl Subject: Re: [OSL | CCIE_Voice] SIP Phones in CME description in voice register pool config mode. On Sat, Oct 2, 2010 at 10:51 AM, CCIE Voice GMAIL givemeccievoice2...@gmail.com wrote: Hi everyone, I am having a hard time remembering what command will affect the number displayed in the upper-right of the phones for CME. With SCCP, I know the description command will effect that number. How do you change this value for SIP phones registered to CME? Thanks for the help, Jeff ___ 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] SIP Phones in CME
Make sure after each change you commit to your SIP phones configuration files you do a create profile under voice reg global. Verify you have your config file ready for download by issueing show voice register tftp and check the mac address of your SIP phone is in the list. A couple of other things to note: You need IP connectivity to your CME router. If your phones are remote to you you need to bind the SIP interface you are sourcing your SIP packets and use that IP address as your source address under voice register global. Authenticate register is a must also if your phones are remote to your SIP CME router. Lastly ensure your voice reg pools have username and password and that you have a number assigned to your phones as the primary number. After all these create profile and if still with troubles please post your configs. Cheers On Sat, Oct 2, 2010 at 12:08 PM, CCIE Voice GMAIL givemeccievoice2...@gmail.com wrote: Hi Dan, Thanks for the response. Before I could test what you had told me, I must have screwed something up. I keep getting a message on both of my CME phones saying “Unprovisioned”. I have reloaded my router and re-configured everything again but I am still getting that message. Has anyone seen this before? Jeff *From:* Daniel Berlinski [mailto:dberlin...@gmail.com] *Sent:* Friday, October 01, 2010 2:55 PM *To:* CCIE Voice GMAIL *Cc:* osl osl *Subject:* Re: [OSL | CCIE_Voice] SIP Phones in CME description in voice register pool config mode. On Sat, Oct 2, 2010 at 10:51 AM, CCIE Voice GMAIL givemeccievoice2...@gmail.com wrote: Hi everyone, I am having a hard time remembering what command will affect the number displayed in the upper-right of the phones for CME. With SCCP, I know the description command will effect that number. How do you change this value for SIP phones registered to CME? Thanks for the help, Jeff ___ 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] RES: +Dialing to Site C without Gatekeeper
Its normal Behavior, no worries. De: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] Em nome de ayman labib Enviada em: sexta-feira, 1 de outubro de 2010 18:37 Para: ccie_voice@onlinestudylist.com Assunto: [OSL | CCIE_Voice] +Dialing to Site C without Gatekeeper Hello Experts, I'm working on TEHO to Site C with Gatekeeper. Everything works except for the + sign. On the debug ISDN Q931 on both sites, I see the + being sent and received, but on the phone it only shows my 11 digits (12123945003), but on the bottom of the phone it shows my plus sign. Any ideas??? Thanks in advance CheersErro! O nome de arquivo não foi especificado. Nenhum vírus encontrado nessa mensagem recebida. Verificado por AVG - www.avgbrasil.com.br Versão: 9.0.856 / Banco de dados de vírus: 271.1.1/3170 - Data de Lançamento: 10/01/10 03:34:00 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP Phones in CME
Hi again, I actually reloaded my router with clean configuration and then re-configured CME, however I am still seeing the same problem. I erased the configurations on the phones before this all happened, so I assume this is maybe part of the problem. I dont know why it would be though, as the phones are getting IP addresses from DHCP and communicating with CME. This is my relevant configs: - DHCP FOR PHONES - ip dhcp excluded-address 10.5.202.1 ip dhcp pool SC_PHONES network 10.5.202.0 255.255.255.0 option 150 ip 10.5.202.1 default-router 10.5.202.1 - VOICE SERVICE - voice service voip allow-connections sip to sip fax protocol cisco sip bind control source-interface Vlan250 bind media source-interface Vlan250 registrar server expires max 1200 min 500 - VOICE CODEC - voice class codec 1 codec preference 1 g711alaw codec preference 2 g711ulaw codec preference 3 g729r8 - SIP CME CONFIG - voice register global mode cme source-address 10.5.202.1 port 5060 max-dn 20 max-pool 2 load 7945 SIP45.9-0-3S load 7942 SIP42.9-0-3S authenticate register date-format Y/M/D voicemail 4500 url directory http://10.5.202.1/localdirectory create profile sync 0001302544054013 ntp-server 10.5.200.1 mode directedbroadcast - PHONE 1 LINE 1 - voice register dn 1 number 4001 call-forward b2bua busy 4500 call-forward b2bua noan 4500 timeout 10 allow watch name Site C Phone 1 label 4001 - PHONE 2 LINE 1 - voice register dn 2 number 4002 call-forward b2bua busy 4500 call-forward b2bua noan 4500 timeout 10 allow watch name Site C Phone 2 label 4002 - PHONE 1 (7942) - voice register pool 1 id mac 0024.9733.6C28 type 7942 number 1 dn 1 presence call-list dtmf-relay rtp-nte sip-notify voice-class codec 1 username scuser1 password cisco description +442321314001 blf-speed-dial 1 4002 label SCPH2 4002 device privacy off - PHONE 2 (7945) - voice register pool 2 id mac 0024.14B2.F542 type 7945 number 1 dn 2 presence call-list dtmf-relay rtp-nte sip-notify voice-class codec 1 username scuser2 password cisco description +442321314002 blf-speed-dial 1 4001 label SCPH1 4001 device privacy off - TFTP FILES - tftp-server flash:SIP/apps42.9-0-3TH1-22.sbn alias apps42.9-0-3TH1-22.sbn tftp-server flash:apps42.9-0-3TH1-22.sbn alias cnu42.9-0-3TH1-22.sbn tftp-server flash:SIP/cvm42sip.9-0-3TH1-22.sbn alias cvm42sip.9-0-3TH1-22.sbn tftp-server flash:SIP/dsp42.9-0-3TH1-22.sbn alias dsp42.9-0-3TH1-22.sbn tftp-server flash:SIP/jar42sip.9-0-3TH1-22.sbn alias jar42sip.9-0-3TH1-22.sbn tftp-server flash:SIP/SIP42.9-0-3S.loads alias SIP42.9-0-3S.loads tftp-server flash:SIP/term42.default.loads alias term42.default.loads tftp-server flash:SIP/apps45.9-0-3TH1-22.sbn alias apps45.9-0-3TH1-22.sbn tftp-server flash:SIP/cnu45.9-0-3TH1-22.sbn alias cnu45.9-0-3TH1-22.sbn tftp-server flash:SIP/cvm45sip.9-0-3TH1-22.sbn alias cvm45sip.9-0-3TH1-22.sbn tftp-server flash:SIP/dsp45.9-0-3TH1-22.sbn alias dsp45.9-0-3TH1-22.sbn tftp-server flash:SIP/jar45sip.9-0-3TH1-22.sbn alias jar45sip.9-0-3TH1-22.sbn tftp-server flash:SIP/SIP45.9-0-3S.loads alias SIP45.9-0-3S.loads tftp-server flash:SIP/term45.default.loads alias term45.default.loads - PHONE PORTS ---à interface FastEthernet0/2/2 switchport access vlan 150 switchport voice vlan 250 spanning-tree portfast ! interface FastEthernet0/2/3 switchport access vlan 150 switchport voice vlan 250 spanning-tree portfast I am still seeing the phone say unprovisioned. As you suggested I looked at the show voice register tftp command and I can see the SEPmac.cnf.xml statements for the phones. - show voice register tftp - R3#show voice register tftp tftp-server syncinfo.xml url system:/cme/sipphone/syncinfo.xml tftp-server SIPDefault.cnf url system:/cme/sipphone/SIPDefault.cnf tftp-server softkeyDefault_kpml.xml url system:/cme/sipphone/softkeyDefault_kpml.xml tftp-server softkeyDefault.xml url system:/cme/sipphone/softkeyDefault.xml tftp-server SEP002497336C28.cnf.xml url system:/cme/sipphone/SEP002497336C28.cnf.xml tftp-server SEP002414B2F542.cnf.xml url system:/cme/sipphone/SEP002414B2F542.cnf.xml Also, when I look at the status messages on the phone, I see a Error Verifying Config Info message. Any help is appreciated, Jeff From: Daniel Berlinski [mailto:dberlin...@gmail.com] Sent: Friday, October 01, 2010 4:20 PM To: CCIE Voice GMAIL Cc: osl osl Subject: Re: [OSL | CCIE_Voice] SIP Phones in CME Make sure after each change you commit to your SIP phones configuration files you do a create profile under voice reg global. Verify you have your config file ready for download by issueing show voice register tftp and check the
Re: [OSL | CCIE_Voice] SIP Phones in CME
I beleive you are missing tftp path flash: under voice register global. Can you try, create profile and let us know? On Sat, Oct 2, 2010 at 1:11 PM, CCIE Voice GMAIL givemeccievoice2...@gmail.com wrote: Hi again, I actually reloaded my router with clean configuration and then re-configured CME, however I am still seeing the same problem. I erased the configurations on the phones before this all happened, so I assume this is maybe part of the problem. I don’t know why it would be though, as the phones are getting IP addresses from DHCP and communicating with CME. This is my relevant configs: - DHCP FOR PHONES - ip dhcp excluded-address 10.5.202.1 ip dhcp pool SC_PHONES network 10.5.202.0 255.255.255.0 option 150 ip 10.5.202.1 default-router 10.5.202.1 - VOICE SERVICE - voice service voip allow-connections sip to sip fax protocol cisco sip bind control source-interface Vlan250 bind media source-interface Vlan250 registrar server expires max 1200 min 500 - VOICE CODEC - voice class codec 1 codec preference 1 g711alaw codec preference 2 g711ulaw codec preference 3 g729r8 - SIP CME CONFIG - voice register global mode cme source-address 10.5.202.1 port 5060 max-dn 20 max-pool 2 load 7945 SIP45.9-0-3S load 7942 SIP42.9-0-3S authenticate register date-format Y/M/D voicemail 4500 url directory http://10.5.202.1/localdirectory create profile sync 0001302544054013 ntp-server 10.5.200.1 mode directedbroadcast - PHONE 1 LINE 1 - voice register dn 1 number 4001 call-forward b2bua busy 4500 call-forward b2bua noan 4500 timeout 10 allow watch name Site C Phone 1 label 4001 - PHONE 2 LINE 1 - voice register dn 2 number 4002 call-forward b2bua busy 4500 call-forward b2bua noan 4500 timeout 10 allow watch name Site C Phone 2 label 4002 - PHONE 1 (7942) - voice register pool 1 id mac 0024.9733.6C28 type 7942 number 1 dn 1 presence call-list dtmf-relay rtp-nte sip-notify voice-class codec 1 username scuser1 password cisco description +442321314001 blf-speed-dial 1 4002 label SCPH2 4002 device privacy off - PHONE 2 (7945) - voice register pool 2 id mac 0024.14B2.F542 type 7945 number 1 dn 2 presence call-list dtmf-relay rtp-nte sip-notify voice-class codec 1 username scuser2 password cisco description +442321314002 blf-speed-dial 1 4001 label SCPH1 4001 device privacy off - TFTP FILES - tftp-server flash:SIP/apps42.9-0-3TH1-22.sbn alias apps42.9-0-3TH1-22.sbn tftp-server flash:apps42.9-0-3TH1-22.sbn alias cnu42.9-0-3TH1-22.sbn tftp-server flash:SIP/cvm42sip.9-0-3TH1-22.sbn alias cvm42sip.9-0-3TH1-22.sbn tftp-server flash:SIP/dsp42.9-0-3TH1-22.sbn alias dsp42.9-0-3TH1-22.sbn tftp-server flash:SIP/jar42sip.9-0-3TH1-22.sbn alias jar42sip.9-0-3TH1-22.sbn tftp-server flash:SIP/SIP42.9-0-3S.loads alias SIP42.9-0-3S.loads tftp-server flash:SIP/term42.default.loads alias term42.default.loads tftp-server flash:SIP/apps45.9-0-3TH1-22.sbn alias apps45.9-0-3TH1-22.sbn tftp-server flash:SIP/cnu45.9-0-3TH1-22.sbn alias cnu45.9-0-3TH1-22.sbn tftp-server flash:SIP/cvm45sip.9-0-3TH1-22.sbn alias cvm45sip.9-0-3TH1-22.sbn tftp-server flash:SIP/dsp45.9-0-3TH1-22.sbn alias dsp45.9-0-3TH1-22.sbn tftp-server flash:SIP/jar45sip.9-0-3TH1-22.sbn alias jar45sip.9-0-3TH1-22.sbn tftp-server flash:SIP/SIP45.9-0-3S.loads alias SIP45.9-0-3S.loads tftp-server flash:SIP/term45.default.loads alias term45.default.loads - PHONE PORTS ---à interface FastEthernet0/2/2 switchport access vlan 150 switchport voice vlan 250 spanning-tree portfast ! interface FastEthernet0/2/3 switchport access vlan 150 switchport voice vlan 250 spanning-tree portfast I am still seeing the phone say unprovisioned. As you suggested I looked at the show voice register tftp command and I can see the SEPmac.cnf.xml statements for the phones. - show voice register tftp - R3#show voice register tftp tftp-server syncinfo.xml url system:/cme/sipphone/syncinfo.xml tftp-server SIPDefault.cnf url system:/cme/sipphone/SIPDefault.cnf tftp-server softkeyDefault_kpml.xml url system:/cme/sipphone/softkeyDefault_kpml.xml tftp-server softkeyDefault.xml url system:/cme/sipphone/softkeyDefault.xml tftp-server SEP002497336C28.cnf.xml url system:/cme/sipphone/SEP002497336C28.cnf.xml tftp-server SEP002414B2F542.cnf.xml url system:/cme/sipphone/SEP002414B2F542.cnf.xml Also, when I look at the status messages on the phone, I see a “Error Verifying Config Info” message. Any help is appreciated, Jeff *From:* Daniel Berlinski [mailto:dberlin...@gmail.com] *Sent:* Friday, October 01, 2010 4:20 PM *To:* CCIE Voice
[OSL | CCIE_Voice] Voice Lab Equipment on Sale
Hi guys, Now that I'm done with my lab, I have the following voice lab equipment on sale. Please let me know if you are interested. Thanks. 1 x 3640 router 2 x 2811 router 3550 24 port POE switch 2509 router (Access Server) 2522 router (FR Switch) AIM-CUE HWIC-4ESW PVDM2-48 x 1 PVDM2-16 x 1 6 x E1 cards (both 1 and 2 port) IP phone 7961 x 1 IP phone 7970 x 1 IP phone 7960 x 3 Regards, Duke ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Single Number Reach
Is there a specific setting to force the ip phone to display an in use message in the event the pstn phone answers the incoming call? On Oct 1, 2010, at 11:42 AM, Graham Hopkins wrote: Same here , I was beginning to think that no patterns are matched in calling number transformations - but I tested with a pattern of ! and a mask of 12345 and that works. So it would appear that there is a mismatch between \+1480.! and the calling number, which does seem odd as if you leave it alone it gets sent to the PSTN as +1480XXX. It would appear that it should match as the pattern ! with XXX works, but as Mark says this doesn't do what he requires Graham On 1 Oct 2010, at 19:23, Mark Holloway wrote: The only issue with this is you don't know if the calling party is Subscriber, National, or International, so you can't use XXX because if BR2 or BR1 calls HQ3 the From number would only show the first 7 digits. On Oct 1, 2010, at 11:21 AM, sisiaji wrote: yeah, you are right, I was referring to RP/RL transformations... i tested it and i got the same in my lab so i guess, as you already mentioned before, the way to do it is to actually put Calling Party Transform Mask to be XXX on the RL (for RG member). On Fri, Oct 1, 2010 at 7:52 PM, Mark Holloway m...@markholloway.com wrote: When doing it under Call Routing Transformation Pattern Calling Party Transformation you have to use \+ When doing it on the Calling Party transform mask on a Route Pattern or Route List you don't use \ On Oct 1, 2010, at 10:44 AM, sisiaji wrote: calling party transformation is done without prefix \ On Fri, Oct 1, 2010 at 7:00 PM, Mark Holloway m...@markholloway.com wrote: The crazy thing is I tried this but I couldn't get it to work. PT_HQ_CALLING_OUT assigned to CSS_HQ_CALLING_OUT assigned to CALLING Number Transform on the Outbound portion of the HQ gateway. Calling Party Transformation: PT_HQ_CALLING_OUT pattern is \+1480.! (replace 480 with what your HQ area code is) Strip Predot That should make the outbound From number +14805552001 appear as 5552001 on the PSTN phone. and I should see 5552001 in the isdn q931 debug output. I'm still seeing the full E164 number. On Oct 1, 2010, at 9:22 AM, Graham Hopkins wrote: Well I'm just showing the full E.164 as that's what the lab I'm looking at looks for. However I guess you could strip the HQ area code at the gateway with the calling party transformation. In the real world (plan to visit that soon) then the remote destination is likely to be a mobile phone which isn't really local to any gateway - at least not here in the UK so would be a national call from anywhere. Graham On 1 Oct 2010, at 17:10, Mark Holloway wrote: Sorry, I meant Translation Patterns, not Profiles. Still working on the From number presentation. I'm assuming that if HQ1 calls HQ3 the PSTN phone should show a 7 digit From number, but if BR1 calls 2003 the PSTN should show a 10 digit From number. Would you guys agree? On Oct 1, 2010, at 8:56 AM, Mark Holloway wrote: Graham, same thing here. This is a summary of what I've done to get it working correctly. I eliminated using Translation Profiles as I didn't find them necessary for this. Create PT_SNR which is assigned to CSS_SNR Create a Remote Destination Profile and assign CSS_SNR to both Calling Search Space and Rerouting Calling Search Space. Build/associate your end user with this Remote Destination Profile. Build a Route List (RL_SNR) that includes just the HQ gateway and set the Calling Party External Phone Mask to On. Doing this in the Route Pattern won't work. Set Called Party to Subscriber (assuming the Remote Destination number is a local number). Lastly, build a Route Pattern that matches your Remote Destination Profile external number and assign it to PT_SNR and RL_SNR. The only thing about this method is that when calls from 2001 ring 2003 which rings the PSTN, this method is using the external mask which means HQ1's external mask is E164. Typically when a Subscriber call egresses the HQ gateway you would want the From number to be 7 digits. Are you guys putting a Calling Party Transformation on your HQ gateway to strip off the HQ area code for Subscriber calls? For all other purposes of presenting 7, 10, or E164, I have always used the Calling Party Transform in either the Route Pattern or Route List's Route Group. Thanks, Mark On Oct 1, 2010, at 7:25 AM, Graham Hopkins wrote: Just hit the same problem in Vol2 Lab4 and I can confirm that this doesn't work at the RP level but does work at the RL level. Is this a known bug ? Graham On 1 Oct 2010, at 13:35, Tam Nhu wrote: Hi Mark, The EPNM does not work at RP for SNR. Have you try to set EPNM at RL level? TN.
