[OSL | CCIE_Voice] MVA partial match issue
hi, I config the Service parameter for MVA , using partial match 7 digit . However when I dial the RD using 7 digit ,it never works. seem like UCM only take Full match. I heard this is bug, Any suggestion for the work around if still want to use partial match ? d On Sun, Mar 17, 2013 at 9:58 AM, michael.se...@compucom.com wrote: Greetings, I think you are doing everything right just need a few tweaks. Place a call from the PSTN line 2 to 3033300 and do debug isdn q931 on gateway. What digits do you see for the calling number. 7 or 10? If seeing 7 digits inbound change your Remote Destination Number to 525, without the 9. If you are seeing 10 digits inbound the NPA, NXX, TNTN change your remote destination number to XXX525, in other words match what you're seeing in the isdn debug for calling party and make that you're Remote Destination Number. Do NOT require the prefix of 9 on the Remote Destination Number. Also, under Remote Destination Information make sure you are putting a tick in Mobile Phone checkbox and a tick in the Enable Mobile Connect checkbox. Otherwise your configuration looks good. Hope you find this helpful. Michael Sears CCIE 38404 Date: Sun, 17 Mar 2013 18:23:01 +0530 From: sanity insanity networksanitytoinsan...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Mobile Voice Access not working since many days!! Message-ID: cag4zmyxmd5xj67pwv+_gpabedjoydyg+zbnmugtsfuj3nsx...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hello All, I have been trying this config for MVA for close to 2 weeks now and it does not work . Here are the details The Issue : == I am trying to Intiate a Call from PSTN phone to site B gateway (H323) 3033300 it should ask for authentication once authenticated press 1 to make any 4 digit calls if it is from SB phone 1 . Make sure to display 4 digits number for calling number along with calling name SB Phone 1 they can use local gateway to make the call. Also 2nd line on PSTN phone should be used to dial 3033300 and you will prompted to login. Details: = My config is following 1) The dial-peers are set in the following way dial-peer voice 102 voip preference 2 destination-pattern 3300 session target ipv4:ip address of the CUCM Pub dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 5 pots service cmm incoming called-number 3300 no digit-strip 2) here is the MVA service url ! application service cmm http://ip address of the CUCM Pub:8080/ccmivr/pages/IVRMainpage.vxml ! 3) I am stripping 3033300 coming from pstn to last 4 digits using a translation-rule on the voice-port level . That is 3033300 becomes 3300 when it reaches CUCM. 4) On CUCM in the service parameters... Enable Mobile Voice access is set to True Mobile voice access number is 3300 Matching caller id with Remote Destination is Partial Match Number of digits of Caller ID Partial Match is 7 5) The Mobility softkey has been added for on hold and connected at the softkey template level and applied to the phone ( SB PH1) 6)At the User SB phone 1 I have enabled Enable Mobility and Enable Mobile Voice Access also selected the MAC address of the phone 7) Created a Remote Dest profile and selected user id of sb ph1 and the correct calling search space for the phone 8) Added a Remoted Destination number of 9525 9) Also went to device phone and selected the Owner User ID of SB Ph1 10) Cisco Unified Mobile Voice Access Service is running on both Sub and Pub on CUCM Questions : 1) I now dial from the pstn line 9525 on the pstn phone to 3033300 . The prompt I get asks me for a pin . I enter 1 and the call drops . I Even tried entering 12345 ( which the pin for user SB Phone 1) and still the call drops after the prompt. Anything wrong the above config? Anything missing in the config ? Any suggestions? 2) I am strip the number to last 4 digits ( as in step 3) . Is this correct procedure? 3) There is also no QOS setup in the config for now . Anything related to Bandwidth here? Please help! -MJ ___ For more information regarding industry leading 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] help on cue
hi guys, call to users vm works just fine but when transfering to a vm extension it drops. pls what could be the problem?? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CME-SRST
: text/plain; charset=iso-8859-1 Hi, thanks for your help yes I have the type on the ephone, the way that I found to fix the problem is copyin the configuraton on notepad, removing the telephony-service and putting back again the configuration in the router, the line appears again on the phone and the ephone template takes effect. 2013/3/24 Steve Keller skeller...@gmail.com Bug you do not want to reset phones in SRST mode. It is for this reason that it would be recommended to test your SRST early in the lab , write mem, reload and not do SRST again. CME as SRST was new in these releases and is known to be very buggy. On Sunday, March 24, 2013, Ivan Dar?o S?nchez Calder?n wrote: Hi, When I configure CME-SRST and I apply an ephone-template to a phone, I reset the phone and when the pone comes back there is no line on the phone, I check the configuration and everithing is fine , the button is associated with the phone, this is a bug? someone knows what is the workaround? Thanks -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20130324/8f7fd1d7/attachment-0001.html -- Message: 5 Date: Mon, 25 Mar 2013 01:35:23 +0300 From: CCIEing aboaz...@gmail.com To: ikizoo4 kwon ikiz...@hotmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Outgoing Calls via T1 failed Message-ID: caehpkk+rkt6dq7ezt1ndkaqpzrthteapvwc_8s6yzuid+bz...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi, Changing the selection to top down solve the issue :) thanks On Sat, Mar 23, 2013 at 2:27 AM, ikizoo4 kwon ikiz...@hotmail.com wrote: use Topdwon for channel selection order in GW -- Date: Sat, 23 Mar 2013 00:52:10 +0300 From: aboaz...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Outgoing Calls via T1 failed Hi all, After configuring my HQ GW as MGCP, then configure my T1 to register with cucm , I was govern by the lack of the DSP resources, which force me to define only 16 channel out of the 23 on my pri-group under the T1 controller configuration !! here is the config : controller T1 0/0/0 pri-group timeslots 1-16 service mgcp Then I faced a problem with my outgoing calls , the calls was dropping due to the cause *Requested circuit/channel not available* My Question here, as there is 16 channel in my Pri-group are already configured, why all calls get dropped with cause of non availability of the resources, Why not to use one of the available channels (1-16) Appreciate your help here is below the output of debug q931 for one of my outgoing calls: *SDN Se0/0/0:23 Q931: TX - SETUP pd = 8 callref = 0x000F * *Bearer Capability i = 0x8090A2 * *Standard = CCITT * *Transfer Capability = Speech * *Transfer Mode = Circuit * *Transfer Rate = 64 kbit/s * *Channel ID i = 0xA98390 * *Exclusive, Channel 16 * *Display i = 'HQ PH1' * *Calling Party Number i = 0x2181, '7772022001' * *Plan:ISDN, Type:National * *Called Party Number i = 0x81, '911' * *Plan:ISDN, Type:Unknown* *ISDN Se0/0/0:23 Q931: RX - RELEASE_COMP pd = 8 callref = 0x800F * *Cause i = 0x82AC1810 - Requested circuit/channel not available* ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20130325/54e5c4f5/attachment.html -- ___ CCIE_Voice mailing list CCIE_Voice@onlinestudylist.com http://onlinestudylist.com/mailman/listinfo/ccie_voice End of CCIE_Voice Digest, Vol 85, Issue 89 ** ___ For more information regarding industry leading 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 partial match issue
You can have the rd with 7 digits only and without the 9 for pstn access - use either application dial rules (match 7 digits, prefix 9) or a translation pattern to modify the rd to match your existing local route pattern. I'm not sure if there's an MVA bug in this version of cucm, but its pretty easy to configure it so that you always have a full match since you will likely have only one rd. This is what I do for the lab. A real world (for nanp) example of MVA partial match would be using e164 address for all rd (+1 npa-nxx-) and set partial match to 10 or 7 depending on whether all sites receive inbound ani as 10d for local calls or if any sites receives only 7d. This would also work for lab, but takes extra steps if you aren't already required to use + dialing For partial match to work, the rd must be longer than the inbound ani (ani 7d and rd +11d). You cannot use partial match with an ani longer than the rd (ani 10d and rd 7d), in this case your options would be to apply inbound transformation on the gateway to make rd ani shorter (ie match the rd) or make your rd longer and manipulate outbound dnis to make it route. On Mar 25, 2013 2:00 AM, donny f f.faraday...@gmail.com wrote: hi, I config the Service parameter for MVA , using partial match 7 digit . However when I dial the RD using 7 digit ,it never works. seem like UCM only take Full match. I heard this is bug, Any suggestion for the work around if still want to use partial match ? d On Sun, Mar 17, 2013 at 9:58 AM, michael.se...@compucom.com wrote: Greetings, I think you are doing everything right just need a few tweaks. Place a call from the PSTN line 2 to 3033300 and do debug isdn q931 on gateway. What digits do you see for the calling number. 7 or 10? If seeing 7 digits inbound change your Remote Destination Number to 525, without the 9. If you are seeing 10 digits inbound the NPA, NXX, TNTN change your remote destination number to XXX525, in other words match what you're seeing in the isdn debug for calling party and make that you're Remote Destination Number. Do NOT require the prefix of 9 on the Remote Destination Number. Also, under Remote Destination Information make sure you are putting a tick in Mobile Phone checkbox and a tick in the Enable Mobile Connect checkbox. Otherwise your configuration looks good. Hope you find this helpful. Michael Sears CCIE 38404 Date: Sun, 17 Mar 2013 18:23:01 +0530 From: sanity insanity networksanitytoinsan...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Mobile Voice Access not working since many days!! Message-ID: cag4zmyxmd5xj67pwv+_gpabedjoydyg+zbnmugtsfuj3nsx...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hello All, I have been trying this config for MVA for close to 2 weeks now and it does not work . Here are the details The Issue : == I am trying to Intiate a Call from PSTN phone to site B gateway (H323) 3033300 it should ask for authentication once authenticated press 1 to make any 4 digit calls if it is from SB phone 1 . Make sure to display 4 digits number for calling number along with calling name SB Phone 1 they can use local gateway to make the call. Also 2nd line on PSTN phone should be used to dial 3033300 and you will prompted to login. Details: = My config is following 1) The dial-peers are set in the following way dial-peer voice 102 voip preference 2 destination-pattern 3300 session target ipv4:ip address of the CUCM Pub dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 5 pots service cmm incoming called-number 3300 no digit-strip 2) here is the MVA service url ! application service cmm http://ip address of the CUCM Pub:8080/ccmivr/pages/IVRMainpage.vxml ! 3) I am stripping 3033300 coming from pstn to last 4 digits using a translation-rule on the voice-port level . That is 3033300 becomes 3300 when it reaches CUCM. 4) On CUCM in the service parameters... Enable Mobile Voice access is set to True Mobile voice access number is 3300 Matching caller id with Remote Destination is Partial Match Number of digits of Caller ID Partial Match is 7 5) The Mobility softkey has been added for on hold and connected at the softkey template level and applied to the phone ( SB PH1) 6)At the User SB phone 1 I have enabled Enable Mobility and Enable Mobile Voice Access also selected the MAC address of the phone 7) Created a Remote Dest profile and selected user id of sb ph1 and the correct calling search space for the phone 8) Added a Remoted Destination number of 9525 9) Also went to device phone and selected the Owner User ID of SB Ph1 10) Cisco Unified Mobile Voice Access Service is running on both Sub and Pub on CUCM Questions : 1) I now dial from the pstn line 9525 on the pstn phone to
Re: [OSL | CCIE_Voice] UCCX agent routing and script verification!!
hi William, Thanks for your reply. I have now added the following for step (b) and added (c) ,(d),(e) -start -accept -play prompt ( welcome prompt) -menu( triggering contact , operator.wav) -option 1:- a) call redirect to 4001 b) If successful terminate c) if busy goto queueloop d) if Invalid goto queueloop e) if Unsuccessful goto queueloop - Under Select Resource ( triggering contact - from CSQ) :- a)queueLoop: 1) Now when I call 4000 it says Thank you for calling this number ...if you dialled this number by mistake please press 1 else someone will be with you shortly Tests done:- i I press 1 it goes to 4001 correctly - This works ii If I don't press any key and wait for timeout the same prompts I hear with are u still there ?4 times and then it goes to the agents 4101 and 4102 - not clear whether this is right ii If I press any other key other than 1 it says please dial again and I need to press the same key ( for example digit 3 on the keypad) atleast 3 times before it goes to the queue - not sure if this is the correct method. Please let me know if this is correct? Thanks once again. -Mj On Fri, Mar 22, 2013 at 12:59 AM, William Bell b...@ucguerrilla.com wrote: 1) when I call 4000 I can hear the greeting saying Press 1 to be transferred to priority agent or stay online for next available agent . The call does not ring on the ipcc agent phones of 4101 and 4102 but when I press 1 it is transferred to 4001 as expected. My question is what is preventing it from ringing 4101 and 4102 even though the agents are in a Ready state? Given the way you presented your script logic this behavior is expected. You are asking the contact to press 1 and handling the transfer action prior to the Select Resource step. 2) The resources set for 4101 and 4102 are in Resource group name S and the CSQ for this is named as CSQ. The resource criteria is Longest Available. Is this correct? Longest Idle == Longest Available 3) Any other parameter that needs to be checked under the Resource group or the CSQ? Can't say. Assuming you have configured your resources and CSQ correctly and you have properly employed either Resource Group or Skills based routing then I think you are OK. If you have failed to configure resources/CSQ/etc. correctly then you are not OK. 4)Is the configuration steps correct ? What steps are missing if any and how do we correct it? Is the script correct? Is something not behaving the way you want or expect it to? If yes, then something is provisioned incorrectly. Your script has a logic flaw. -option 1:- a) call redirect to 4001 b) If successful goto queueLoop Step (b) doesn't make sense to me. If you successfully redirect the contact then the script logic shouldn't go to the queueLoop. You should terminate. -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 21, 2013, at 1:03 PM, sanity insanity wrote: Hi All, Need your help. I am configuring DNs 4101 4102 ( both DNs are uccx agent extensions). Calls to 4000 should here a greeting Press 1 to be transferred to priority agent or stay online for next available agent . If the caller presses 1 , calls should be transferred to 4001. Otherwise it should be hunted as per Longest idle time. These are the configuration steps I followed -- 1) recorded a prompt for the greeting called operator.wav 2) Configured one button login for the phone dns ( agent DNs - 4101 4102) 3) Setup the CSQ and resources in UCCX 4) Wrote the following script... -start -accept -play prompt ( welcome prompt) -menu( triggering contact , operator.wav) -option 1:- a) call redirect to 4001 b) If successful goto queueLoop - Under Select Resource ( triggering contact - from CSQ) :- a)queueLoop: -End 5) Configured a trigger for 4000 Questions : 1) when I call 4000 I can hear the greeting saying Press 1 to be transferred to priority agent or stay online for next available agent . The call does not ring on the ipcc agent phones of 4101 and 4102 but when I press 1 it is transferred to 4001 as expected. My question is what is preventing it from ringing 4101 and 4102 even though the agents are in a Ready state? 2) The resources set for 4101 and 4102 are in Resource group name S and the CSQ for this is named as CSQ. The resource criteria is Longest Available. Is this correct? 3) Any other parameter that needs to be checked under the Resource group or the CSQ? 4)Is the configuration steps correct ? What steps are missing if any and how do we correct it? Is the script correct? - MJ ___ For more information regarding industry leading 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
[OSL | CCIE_Voice] [Translation-Pattern and Voice Translation Profile ]
Hi Folks, dialplan-pattern 1 902312... extension-length 4... when it comes to voice translation rule is it mandatory to remove above command. Thanks, Ravi. ___ For more information regarding industry leading 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 partial match issue
hi justin, the full match, always work. Just when we do partial , it never , and always ask for RD number after we dial the MVA. I did exactly you suggest below. - A real world (for nanp) example of MVA partial match would be using e164 address for all rd (+1 npa-nxx-) and set partial match to 10 or 7 depending on whether all sites receive inbound ani as 10d for local calls or if any sites receives only 7d. This would also work for lab, but takes extra steps if you aren't already required to use + dialing is it best practice to always use Full Match, and the we hv to modify using Xfrom on GW and Application Dial Rule/TP to adjust ? tks d On Mon, Mar 25, 2013 at 2:50 AM, Justin Carney justin.s.car...@gmail.comwrote: You can have the rd with 7 digits only and without the 9 for pstn access - use either application dial rules (match 7 digits, prefix 9) or a translation pattern to modify the rd to match your existing local route pattern. I'm not sure if there's an MVA bug in this version of cucm, but its pretty easy to configure it so that you always have a full match since you will likely have only one rd. This is what I do for the lab. A real world (for nanp) example of MVA partial match would be using e164 address for all rd (+1 npa-nxx-) and set partial match to 10 or 7 depending on whether all sites receive inbound ani as 10d for local calls or if any sites receives only 7d. This would also work for lab, but takes extra steps if you aren't already required to use + dialing For partial match to work, the rd must be longer than the inbound ani (ani 7d and rd +11d). You cannot use partial match with an ani longer than the rd (ani 10d and rd 7d), in this case your options would be to apply inbound transformation on the gateway to make rd ani shorter (ie match the rd) or make your rd longer and manipulate outbound dnis to make it route. On Mar 25, 2013 2:00 AM, donny f f.faraday...@gmail.com wrote: hi, I config the Service parameter for MVA , using partial match 7 digit . However when I dial the RD using 7 digit ,it never works. seem like UCM only take Full match. I heard this is bug, Any suggestion for the work around if still want to use partial match ? d On Sun, Mar 17, 2013 at 9:58 AM, michael.se...@compucom.com wrote: Greetings, I think you are doing everything right just need a few tweaks. Place a call from the PSTN line 2 to 3033300 and do debug isdn q931 on gateway. What digits do you see for the calling number. 7 or 10? If seeing 7 digits inbound change your Remote Destination Number to 525, without the 9. If you are seeing 10 digits inbound the NPA, NXX, TNTN change your remote destination number to XXX525, in other words match what you're seeing in the isdn debug for calling party and make that you're Remote Destination Number. Do NOT require the prefix of 9 on the Remote Destination Number. Also, under Remote Destination Information make sure you are putting a tick in Mobile Phone checkbox and a tick in the Enable Mobile Connect checkbox. Otherwise your configuration looks good. Hope you find this helpful. Michael Sears CCIE 38404 Date: Sun, 17 Mar 2013 18:23:01 +0530 From: sanity insanity networksanitytoinsan...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Mobile Voice Access not working since many days!! Message-ID: cag4zmyxmd5xj67pwv+_gpabedjoydyg+zbnmugtsfuj3nsx...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hello All, I have been trying this config for MVA for close to 2 weeks now and it does not work . Here are the details The Issue : == I am trying to Intiate a Call from PSTN phone to site B gateway (H323) 3033300 it should ask for authentication once authenticated press 1 to make any 4 digit calls if it is from SB phone 1 . Make sure to display 4 digits number for calling number along with calling name SB Phone 1 they can use local gateway to make the call. Also 2nd line on PSTN phone should be used to dial 3033300 and you will prompted to login. Details: = My config is following 1) The dial-peers are set in the following way dial-peer voice 102 voip preference 2 destination-pattern 3300 session target ipv4:ip address of the CUCM Pub dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 5 pots service cmm incoming called-number 3300 no digit-strip 2) here is the MVA service url ! application service cmm http://ip address of the CUCM Pub:8080/ccmivr/pages/IVRMainpage.vxml ! 3) I am stripping 3033300 coming from pstn to last 4 digits using a translation-rule on the voice-port level . That is 3033300 becomes 3300 when it reaches CUCM. 4) On CUCM in the service parameters... Enable Mobile Voice access is set to True Mobile voice
Re: [OSL | CCIE_Voice] Network side vs User side clocking !
Since no one responded, I guess I'll be the geek for now. The difference between using network-side clocking and user-side clocking is similar to the DCE / DTE difference. PRI and E1 are implementations of the ISDN protocol over T1 à a point-to-point, synchronous circuit. Synchronous circuits require timing, so one end should operate as the master of the clock and the other end should recover the clock/timing/synchronization from the master. One of the amazing things about T1 design is that one doesn’t necessarily have to setup clocking correctly to get the circuit to work. However, you must configure it correctly to get the circuit to work well. When I started out, clocking was rudely referred to as master/slave, but fortunately many vendors like Cisco adopted more politically-correct terminology. Network = Master (usually telco side). User side is the other end. In Cisco's case the clock master is defined under the controller T1 x/x/x as 'clock source internal.' Usually a PRI handoff from the PSTN in the real world would be set as so for you on whatever equipment they use to provide you your T1 span. I'd expect your controllers on the PSTN router in the lab to be set ‘clock source internal.’ Don't know for sure if Cisco does it that way in the lab; I assume they would if they wanted it to bear any resemblance to the real world. Therefore, the site A,B C controllers would normally be set opposite to network side, which would be to say, ‘clock source line.’ Since this is the default, the command becomes invisible in the config. A good way someone told me to remember this is, You set the controller to ‘clock source line’ if you want it to look up the line for the clock. The part in your question about layer 1 and layer 2 doesn't exactly pertain to clocking. The 4-wire T1 is layer 1, pins 1-2/4-5. Q.921 is layer 2. Q.931 is layer 3. Without going into much detail, what you can do with the network-clock-participate and network-clock-select command is to inform the router about your preferences as to what to do with the clock timing(s) it recovers on its various T1 controllers. For example in the real world, it is possible to have a PRI from carrier X and another span from carrier Y. In this case you might want to use network-clock-select to control the relative priority of the two clock sources. It is a best practice to set network-clock-select explicitly even if you have only one PRI/E1, as doing so avoids slips. Network-clock-participate informs the router that it may be possible to gather a clock source on a particular wic, and the network-clock-select informs the router as to which one of those to use to sync the backplane of the router. Your PVDM2s that provide the DSPs for the PRIs are probably installed on the backplane, so it would probably be nice if they had benefit to the same clock synchronication. This has nothing to do with NTP. Different kind of clock. Use the following command to check your T1 clocking. 2951R2#show network-clocks Network Clock Configuration --- Priority Clock SourceClock State Clock Type 1 T1 0/0/0GOODT1 2 T1 0/1/0GOODT1 10 Backplane GOODPLL Current Primary Clock Source --- Priority Clock SourceClock State Clock Type 1 T1 0/0/0GOODT1 https://supportforums.cisco.com/thread/189145 Thanks On Sat, Mar 23, 2013 at 6:07 PM, CCIEing aboaz...@gmail.com wrote: Hi geeks :) What is the difference between using Network side clocking and User Side clocking. Regarding the exam, do they ask us to use any one of the both in particular ? I saw practice question informing that the PRI circuit layer 2 should be user side where as it will be a network side clocking for layer 1 as for the last sentence (network site), I would assume that we will use *network-clock- participate wic X* * * *Waiting your valuable input * ___ For more information regarding industry leading 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] Network side vs User side clocking !
Really good explanation from Tony. As an addendum to him, you can use show controllers command to verify that the controller is set to clock source line, which, as Tony already mentioned, is hidden being default. On Tue, Mar 26, 2013 at 8:26 AM, Tony Zunt tony.z...@gmail.com wrote: Since no one responded, I guess I'll be the geek for now. The difference between using network-side clocking and user-side clocking is similar to the DCE / DTE difference. PRI and E1 are implementations of the ISDN protocol over T1 à a point-to-point, synchronous circuit. Synchronous circuits require timing, so one end should operate as the master of the clock and the other end should recover the clock/timing/synchronization from the master. One of the amazing things about T1 design is that one doesn’t necessarily have to setup clocking correctly to get the circuit to work. However, you must configure it correctly to get the circuit to work well. When I started out, clocking was rudely referred to as master/slave, but fortunately many vendors like Cisco adopted more politically-correct terminology. Network = Master (usually telco side). User side is the other end. In Cisco's case the clock master is defined under the controller T1 x/x/x as 'clock source internal.' Usually a PRI handoff from the PSTN in the real world would be set as so for you on whatever equipment they use to provide you your T1 span. I'd expect your controllers on the PSTN router in the lab to be set ‘clock source internal.’ Don't know for sure if Cisco does it that way in the lab; I assume they would if they wanted it to bear any resemblance to the real world. Therefore, the site A,B C controllers would normally be set opposite to network side, which would be to say, ‘clock source line.’ Since this is the default, the command becomes invisible in the config. A good way someone told me to remember this is, You set the controller to ‘clock source line’ if you want it to look up the line for the clock. The part in your question about layer 1 and layer 2 doesn't exactly pertain to clocking. The 4-wire T1 is layer 1, pins 1-2/4-5. Q.921 is layer 2. Q.931 is layer 3. Without going into much detail, what you can do with the network-clock-participate and network-clock-select command is to inform the router about your preferences as to what to do with the clock timing(s) it recovers on its various T1 controllers. For example in the real world, it is possible to have a PRI from carrier X and another span from carrier Y. In this case you might want to use network-clock-select to control the relative priority of the two clock sources. It is a best practice to set network-clock-select explicitly even if you have only one PRI/E1, as doing so avoids slips. Network-clock-participate informs the router that it may be possible to gather a clock source on a particular wic, and the network-clock-select informs the router as to which one of those to use to sync the backplane of the router. Your PVDM2s that provide the DSPs for the PRIs are probably installed on the backplane, so it would probably be nice if they had benefit to the same clock synchronication. This has nothing to do with NTP. Different kind of clock. Use the following command to check your T1 clocking. 2951R2#show network-clocks Network Clock Configuration --- Priority Clock SourceClock State Clock Type 1 T1 0/0/0GOODT1 2 T1 0/1/0GOODT1 10 Backplane GOODPLL Current Primary Clock Source --- Priority Clock SourceClock State Clock Type 1 T1 0/0/0GOODT1 https://supportforums.cisco.com/thread/189145 Thanks On Sat, Mar 23, 2013 at 6:07 PM, CCIEing aboaz...@gmail.com wrote: Hi geeks :) What is the difference between using Network side clocking and User Side clocking. Regarding the exam, do they ask us to use any one of the both in particular ? I saw practice question informing that the PRI circuit layer 2 should be user side where as it will be a network side clocking for layer 1 as for the last sentence (network site), I would assume that we will use *network-clock- participate wic X* * * *Waiting your valuable input * ___ For more information regarding industry leading 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 -- Suresh Bhandari
