Re: [OSL | CCIE_Voice] DTMF Issue with SIP TRUNK (CUCM and CUE)
On Thu, Jan 30, 2014 at 8:48 AM, Vignesh Sethuraman sethuvign...@gmail.comwrote: Media Termination Point Required (Checked) MTP Preferred Originating CodecRequired Field: g711ulaw Hi Vignesh, I think if you can set these two to default settings which is MTP Required [uncheck], and MTP Prefered Codec: None, Leave the DTMF Signaling Method to No Preference. Reset the SIP Trunk. You shouldn't need MTP for this operation. Then, if you really want to experiment with MTP insertion, I think you may find this article interesting - http://www.ucguerrilla.com/2012/12/ccie-v-i-shoulda-checked-that-tip-3-mtp.html . Regards, --Somphol. ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] no DTMF Relay to UC via PSTN during HA (testing AAR)
Hi Vignesh, Would it be possible to make a test call from PSTN phone too?Is the result different than call made from SiteB PH2/PH3? Also it might be worth checking the dtmf-relay settings on relevant VoIP dial-peer(s) on SiteA GW too. Regards, --Somphol. On Wed, Jan 22, 2014 at 9:49 PM, Vignesh Sethuraman sethuvign...@gmail.comwrote: Hello Mark, yes, I do have *mgcp dtmf-relay voip codec all mode out-of-band.* Thanks, Viki On Tue, Jan 21, 2014 at 8:57 PM, Mark Thrash (marthras) marth...@cisco.com wrote: Do you have the command Mgcp dtmf codec all out In your mgcp config From: Vignesh Sethuraman sethuvign...@gmail.com Date: Tuesday, January 21, 2014 at 1:51 PM To: ccievoice ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] no DTMF Relay to UC via PSTN during HA (testing AAR) Hello All, Unity Connection not recognizing the password (no DTMF) when the call is routed as following during a high availability situation. SiteB PH2/PH3 --- MGCP T1 Port of SiteB GW My PSTN GW (use to switch call between all sites via pots dialpeers) - SiteA H323 GW - CUCM SUB Unity Connection. * The Unity Connection is playing Message -- Enter you PIN * Unity Connection recognizes SiteB PH2 is a registered user's number , so asks for password * When pressing password unity connection does not recognize that any key is pressed I am facing the same issue as mentioned in the below link but I am using Skinny integration of CUC to CUCM. http://onlinestudylist.com/archives/ccie_voice/2013-August/085101.html Please let me know what I am missing. Thanks, Viki ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] MVA and inbound fast start
On Fri, Nov 15, 2013 at 9:14 AM, Olusegun Oguntuga segunogunt...@gmail.comwrote: Can anyone please explain what exactly needs to be done to get calling name displayed on an enterprise phone when a call is a received via mobile voice access with inbound fast start enabled. Hi Olusegun, I think the best way may be to use slow start for the dial-peer that is pointed to the MVA DN media resource on the H323 GW. So, while the rest of the incoming call are still based on fast start, the incoming call via MVA will use slow start. Regards, --Somphol. ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] MGCP MVA CSS and call routing
Hi StefanoS, The typical CUCM SDI's DETAIL trace with default ticks should be enough. I think it is likely that the CSS applied to RDP doesn't have access to your Mobile Voice Access Directory Number' media resource. (The one you define in Media Resources - Mobile Voice Access),I found that one out from the trace, it should show that the Call Manager try to search for that number and the number is not accessible via the given CSS on the RDP. I read long time ago from an article (which I am sorry I couldn't recall where I see it), it helps me a lot to distinguish between Mobile Voice Access DID and Mobile Voice Access DN. The author wrote that in a cluster where multiple voice gateways spread across area codes or countries, you are likely to have different Mobile Voice Access DID, one for each site. There is however only one Mobile Voice Access DN media resource. So, we can have 5 H323 GW with MVA DID of x1010, x2010, x3010, x4010, and x5010, these are the DN that reach the IVR.Then, you can define a more distinctive extension for Mobile Voice Access DID such as x. [This is the key, once you distinguish these numbers clearly, your trace will be much more easy to understand.I used to assume that the MVA DID and MVA DN must be the same number.] On the H323 Voice Gateway, I also find that you need to allow-connection h323 to h323 for this to work. Below is a bit of an excerpt for what you can see from CUCM SDI trace relevant to MVA operation, you can skip it entirely. Once the PIN (and the RD Number when applicable) is entered via the IVR, and you tried to make the outgoing call by pressing 1, the IVR script will try to establish a call with that Mobile Voice Access DID media resource. (Hence, on every one of those H323 GW, you will need to have a dial-peer that allow x in our case to be reachable from the H323 GW). That's one leg of the outgoing call being made. Another leg is created by that media resource, which I think is a software controlling x, in our case.From the trace below, the work involves searching for a matched owner and DN, etc. Then make a second call leg with the DN. In the trace below, CCM|DbMobility found that Caller ID 258001 is a Remote Destination Number for userId SiteB2. 11/01/2013 15:38:14.871 *CCM|DbMobility: getMatchedRemDest starts: cnumber = 258001* |CLID::StandAloneClusterNID::142.100.64.11LVL::DetailedMASK::ff 11/01/2013 15:38:14.871 CCM|DbMobility: getMatchedRemDest: full match case|CLID::StandAloneClusterNID::142.100.64.11LVL::DetailedMASK::ff 11/01/2013 15:38:14.871 CCM|DbMobility:initRemDest: device pkid [cf26b6d2-64d7-b771-0347-d08f6d8d950c], profile pkid [67569716-698c-f39c-cb7c-c0e60c9c12bf], isDualmode [0], isSmartPhone [0]|CLID::StandAloneClusterNID::142.100.64.11LVL::DetailedMASK::ff 11/01/2013 15:38:14.871 CCM|DbMobilityRemDestTable:initRemDest: initialized a remdest [258001]|CLID::StandAloneClusterNID::142.100.64.11LVL::DetailedMASK::ff 11/01/2013 15:38:14.871 *CCM|DbMobility - RemDest dump: cnumber = 258001*, devicePkid = cf26b6d2-64d7-b771-0347-d08f6d8d950c, remDestProfilePkidStr = 67569716-698c-f39c-cb7c-c0e60c9c12bf, isMobilePhone = 0, isDualMode = 0, isSmartPhone = 0, isSNREnabled= 0, answerTooSoonTimer = 1500, answerTooLateTimer = 19000, delayBeforeRingingCellTimer = 4000, userId = SiteB2, timeZoneIndex = 22, description = 258001, url = |CLID::StandAloneClusterNID::142.100.64.11LVL::DetailedMASK::ff 11/01/2013 15:38:14.871 CCM|DbMobility: found DN association for remdest [258001]|CLID::StandAloneClusterNID::142.100.64.11LVL::DetailedMASK::ff 11/01/2013 15:38:14.871 CCM|DbMobility: found remdest cnumber = 258001, devicepkid = cf26b6d2-64d7-b771-0347-d08f6d8d950c|CLID::StandAloneClusterNID::142.100.64.11LVL::DetailedMASK::ff ... ... 11/01/2013 15:38:14.871 CCM|DbMobilityRemDestTable:initRemDest: initialized a remdest [258001]|CLID::StandAloneClusterNID::142.100.64.11LVL::DetailedMASK::ff 11/01/2013 15:38:14.871 CCM|DbMobility - RemDest dump: cnumber = 258001, devicePkid = cf26b6d2-64d7-b771-0347-d08f6d8d950c, remDestProfilePkidStr = 67569716-698c-f39c-cb7c-c0e60c9c12bf, isMobilePhone = 0, isDualMode = 0, isSmartPhone = 0, isSNREnabled= 0, answerTooSoonTimer = 1500, answerTooLateTimer = 19000, delayBeforeRingingCellTimer = 4000, userId = SiteB2, timeZoneIndex = 22, description = 258001, url = |CLID::StandAloneClusterNID::142.100.64.11LVL::DetailedMASK::ff 11/01/2013 15:38:14.871 CCM|DbMobilityRemDestTable:initMobilityUser: -- created mobility user|CLID::StandAloneClusterNID::142.100.64.11LVL::DetailedMASK::ff 11/01/2013 15:38:14.872 *CCM|DbMobility - User dump: userId = SiteB2*, isIVREnabled = 1, maxDeskPickupWaitTime = 1|CLID::StandAloneClusterNID::142.100.64.11LVL::DetailedMASK::ff ... ... Then, a call is made to the target number. Once the 2nd call leg is established. The two call legs will be joined by an MTP (well and/or Xcoder when applicable). You can see
Re: [OSL | CCIE_Voice] Voice Lab dates all gone?
Hi Frank, There are two seats left on Nov 17 in San Jose at the moment. Regards, --Somphol. On Tue, Nov 5, 2013 at 5:40 AM, Frank Costeira (fcosteir) fcost...@cisco.com wrote: Hi, Has anyone else been able to schedule a date? I don't see any dates passed Nov 10 at RTP or San Jose. Regards, Frank On Oct 30, 2013, at 8:40 PM, Patrick Henderson p.hender...@mac.com wrote: Hi Bill, I read you mail an my heart missed a beat. I just scheduled my lab for Sunday Jan 5th. All the best on the 28th. And good luck with your exam Somphol. Ciao Pat On Oct 30, 2013, at 5:32 PM, Bill Tolentino btolent...@hotmail.com wrote: I installed Chrome and viola! I can see the dates now. I think they did do some re-arranging with the dates today, now allowing weekend testing. In the process, IE Firefox browsers got bugged somehow. In any case, I'm scheduled for Jan 28th now happy to get back to studies! Much thanks Somphol good luck on your exam! Take care! Bill Tolentino -- From: somp...@gmail.com Date: Thu, 31 Oct 2013 11:26:22 +1100 Subject: Re: [OSL | CCIE_Voice] Voice Lab dates all gone? To: btolent...@hotmail.com CC: ccie_voice@onlinestudylist.com It is also very good to see lab slots available on Saturday and Sunday at both RTP San Jose. Weekend lab seem to be for those two locations only at the moment. --Somphol. On Thu, Oct 31, 2013 at 8:43 AM, Somphol Boonjing somp...@gmail.com wrote: On Thu, Oct 31, 2013 at 8:21 AM, Bill Tolentino btolent...@hotmail.com wrote: Then at approximately 7:10am they were all gone. I have been re-checking all day and still no dates for any sites.? Hi Bill, I have just checked.There are still available slots in Bangalore. 17 in Nov/ 6 in Dec/ 36 in Jan. None in Sydney / Brussels / Beijing. San Jose, 1 in Nov, 1 in Dec and 1 in Jan. Tokyo - 8 in Nov, 5 in Jan. RTP - 5 in Nov. Because the back-end of the lab is in San Jose, I think Cisco Cert team can shuffle around to reduce slots in one location and make it appear to another, although not on a daily basis, the lab slots seeing today may drastically change. I was in Sydney last week, there are two equipment sets, but only one is being utilized. And based on that all the available slots are now booked in Sydney. I have also written to Cert Support team to give them the feedback. So far the response is that this is a known issue and too bad just try to book at other locations. My next attempt will be in Bangalore, I hope equipment and the network access speed is bearable. Regards, --Somphol. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com http://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading 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] CTI Application Server (TCP/2748) vs JTAPI Application Server (TCP/2789)
Could anyone help explain or refer me to the documentation that help me understand the role of JTAPI Application Server (tcp/2789) a bit more? I am interested to learn about which application server use that particular port TCP/2789? (CUC / UCCX / CUE / CUPC) I know that both CUE and CUPC (Deskphone mode) and UCCX, all of them, talk to CTI Application Server at port TCP/2748, but does JTAPI Application Serer at TCP/2789 ever get used by any of those application server/client? Note: I find it very confusing when people use rmjtapi account name (in case of UCCX) or cuejtapi (in case of CUE) to talk to CTI Application Server (TCP/2748) which really is a CTI Application Server and is not JTAPI Application (TCP/2789). REF: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/port/9_1_1/CUCM_BK_T2CA6EDE_00_tcp-port-usage-guide-91_chapter_01.htmlhttp://www.cisco.com/en/US/customer/docs/voice_ip_comm/cucm/port/9_1_1/CUCM_BK_T2CA6EDE_00_tcp-port-usage-guide-91_chapter_01.html http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/port/8_0_1/portlist801.html http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/port/7_0/CCM_7.0PortList.pdf Cisco Unified Communications App Unified CM 2748 / TCP CTI application server Cisco Unified Communications App Unified CM 2749 / TCP TLS connection between CTI applications (JTAPI/TSP) and CTIManager Cisco Unified Communications App Unified CM 2789 / TCP JTAPI application server See Also: CUPC Port Usage - http://www.cisco.com/en/US/docs/voice_ip_comm/cupc/7_1/english/release/notes/cupc71.htmlhttp://www.google.com/url?q=http%3A%2F%2Fwww.cisco.com%2Fen%2FUS%2Fcustomer%2Fdocs%2Fvoice_ip_comm%2Fcupc%2F7_1%2Fenglish%2Frelease%2Fnotes%2Fcupc71.htmlsa=Dsntz=1usg=AFrqEzczjzDW2L35ak1yNjFTQ0kPD4lofA UCCX Port Usage - http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_7_0/configuration/guide/uccx70prtuti.pdf CUE Integration Guide that suggests TCP/2748 is used (and there is no reference to TCP/2789 at all) - http://www.cisco.com/en/US/products/sw/voicesw/ps5520/products_configuration_example09186a0080289ef0.shtml ___ For more information regarding industry leading 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] CTI Application Server (TCP/2748) vs JTAPI Application Server (TCP/2789)
On Sun, Nov 3, 2013 at 2:02 AM, Pavan K pav.c...@gmail.com wrote: Taking the example of UCCX, UCCX can sync with ucm and download jtapi libraries from ccm. Its built in jtapi client uses those libraries to communicate with CTI on the ucm server. The term rmjtapi refers to the local credentials used by its jtapi client to connect to CTI. Hi Pavan, Thanks for the information. It is useful with respect to understand how UCCX and UCM communicate. Are we then dealing with the use of the JTAPI in two different contexts? The first context is JTAPI as a layer on top on CTI, in which UCCX talks to CUCM on port TCP/2748 -- the port that is labeled as CTI Application Server. *The second context, a more mysterious one, is the JTAPI Application Server that is listening on CUCM on port TCP/2789.* All Port Usage doco for CUPC / UCCX / CUE refers to TCP/2748 -- 1st meaning of JTAPI. Strictly in the 2nd context, I am still trying to find any documentation or application that use the port to talk to CUCM. There is one clue that I still don't think is accurate is UCCX 9.0.2 Port Utilization guide (PDF page #12 - http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_9_02/configuration/guide/UCCX_BK_P89325D5_00_port-utilization-guide-uccx-902.pdf ) It said JTAPI client listens to TCP/2789 on UCCX utilizing QBE over TCP to talk to CUCM on port TCP/2748. (And the communication is bi-directional) That would depict a tcp packet with SRC=TCP/2789 and DST=TCP/2748. That's not impossible, but even if that is true, still doesn't answer what JTAPI Application Server on CUCM server is for and which application utilize that. What is also important is to be reminded occasionally that UCCX and CUPC (in Deskphone mode) and CUE (integrated with CUCM) do use the same port to talk to CUCM, TCP/2748. Regards, --Somphol. ___ For more information regarding industry leading 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] Voice Lab dates all gone?
On Thu, Oct 31, 2013 at 8:21 AM, Bill Tolentino btolent...@hotmail.comwrote: Then at approximately 7:10am they were all gone. I have been re-checking all day and still no dates for any sites.? Hi Bill, I have just checked.There are still available slots in Bangalore. 17 in Nov/ 6 in Dec/ 36 in Jan. None in Sydney / Brussels / Beijing. San Jose, 1 in Nov, 1 in Dec and 1 in Jan. Tokyo - 8 in Nov, 5 in Jan. RTP - 5 in Nov. Because the back-end of the lab is in San Jose, I think Cisco Cert team can shuffle around to reduce slots in one location and make it appear to another, although not on a daily basis, the lab slots seeing today may drastically change. I was in Sydney last week, there are two equipment sets, but only one is being utilized. And based on that all the available slots are now booked in Sydney. I have also written to Cert Support team to give them the feedback. So far the response is that this is a known issue and too bad just try to book at other locations. My next attempt will be in Bangalore, I hope equipment and the network access speed is bearable. Regards, --Somphol. ___ For more information regarding industry leading 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] Voice Lab dates all gone?
It is also very good to see lab slots available on Saturday and Sunday at both RTP San Jose. Weekend lab seem to be for those two locations only at the moment. --Somphol. On Thu, Oct 31, 2013 at 8:43 AM, Somphol Boonjing somp...@gmail.com wrote: On Thu, Oct 31, 2013 at 8:21 AM, Bill Tolentino btolent...@hotmail.comwrote: Then at approximately 7:10am they were all gone. I have been re-checking all day and still no dates for any sites.? Hi Bill, I have just checked.There are still available slots in Bangalore. 17 in Nov/ 6 in Dec/ 36 in Jan. None in Sydney / Brussels / Beijing. San Jose, 1 in Nov, 1 in Dec and 1 in Jan. Tokyo - 8 in Nov, 5 in Jan. RTP - 5 in Nov. Because the back-end of the lab is in San Jose, I think Cisco Cert team can shuffle around to reduce slots in one location and make it appear to another, although not on a daily basis, the lab slots seeing today may drastically change. I was in Sydney last week, there are two equipment sets, but only one is being utilized. And based on that all the available slots are now booked in Sydney. I have also written to Cert Support team to give them the feedback. So far the response is that this is a known issue and too bad just try to book at other locations. My next attempt will be in Bangalore, I hope equipment and the network access speed is bearable. Regards, --Somphol. ___ For more information regarding industry leading 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] Misleading timer information on BACD documentation page
Just to share some findings on BACD experiment. My BACD is both for the embedded BACD and the external TCL-based BACD (2.1.x.x) running on IOS 12.4(15)T. I always think BACD is fairly straightforward and well-document.And, I have never come close to question the validity of Cisco's own BACD documentation. Step 27 *param* *second-greeting-time**seconds* Example: Router(config-app-param)# param second-greeting-time 45 (Optional) Defines the time delay before the second greeting is played after a caller joins a Cisco Unified CME B-ACD call queue. The same time period is used for the interval between repeats of the second-greeting message. The second greeting is stored in the audio file named en_bacd_allagentsbusy.au. To record a customized second greeting, see the instructions in the Welcome Prompt and Other Audio Files sectionhttp://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html#wp1012403 . •*seconds*—Time interval before the second-greeting message is played or replayed, in seconds. *The range is from 5 to 120. The default is 60.* So, the document is saying, yes, you can set the value of the timer to between 5 and 120, go ahead The debug message is telling me different story however. I am speechless. I sort of know that BACD would be a bit flaky, but to tandem that with misleading/incomplete information on the documentation is just too much. I seriously thought it was a bug on the IOS version I used until I dig up a bit further and found that it is not a bug and I am not the only one who has encountered this. R1#call application voice load x-app-b-acd-aa R1# *Mar 1 00:08:43.951: //-1//HIFS:/hifs_ifs_cb: hifs ifs file read succeeded. size=37673, url=flash:/app-b-acd-aa-2.1.2.3.tcl *Mar 1 00:08:43.951: //-1//HIFS:/hifs_free_idata: hifs_free_idata: 0x667DE330 *Mar 1 00:08:43.955: //-1//HIFS:/hifs_hold_idata: hifs_hold_idata: 0x667DE330 R1# *Mar 1 00:08:51.175: //39//TCL :/tcl_PutsObjCmd: TCL AA: -- max-extension-length is set to default value of 5 -- **Mar 1 00:08:51.179: //39//TCL :/tcl_PutsObjCmd: TCL AA: ++ second-greeting-time is set to less than minimum allowed value of 30 ++* **Mar 1 00:08:51.179: //39//TCL :/tcl_PutsObjCmd: TCL AA: ++ Setting second-greeting-time to minimum value of 30 ++* *Mar 1 00:08:51.179: //39//TCL :/tcl_PutsObjCmd: TCL AA: -- Valid mandatory parameter second-greeting-time = 30 -- *Mar 1 00:08:51.183: //39//TCL :/tcl_PutsObjCmd: TCL AA: -- Valid mandatory parameter call-retry-timer = 20 -- *Mar 1 00:08:51.183: //39//TCL :/tcl_PutsObjCmd: TCL AA: -- Valid mandatory parameter max-time-call-retry = 180 -- *Mar 1 00:08:51.183: //39//TCL :/tcl_PutsObjCmd: TCL AA: -- Valid mandatory parameter max-time-vm-retry = 3 -- *Mar 1 00:08:51.183: //39//TCL :/tcl_PutsObjCmd: TCL AA: -- Valid Mandatory parameter number-of-hunt-grps = 1 -- *Mar 1 00:08:51.187: //39//TCL :/tcl_PutsObjCmd: proc init_perCallvars *Mar 1 00:08:51.187: *Mar 1 00:08:51.187: //39//TCL :/tcl_PutsObjCmd: TCL AA: +++ B-ACD-SERVICE not registered, Starting B-ACD-SERVICE +++ *Mar 1 00:08:51.207: %IVR-6-APP_INFO: TCL B-ACD: B-ACD Service Started *Mar 1 00:08:51.211: //39//TCL :/tcl_PutsObjCmd: TCL B-ACD: B-ACD Service Started *Mar 1 00:08:51.211: //39//TCL :/tcl_PutsObjCmd: TCL B-ACD: Handoff String = x-app-b-acd-aa *Mar 1 00:08:51.239: //39//TCL :/tcl_PutsObjCmd: proc init_perCallvars *Mar 1 00:08:51.239: *Mar 1 00:08:51.259: //39//TCL :/tcl_PutsObjCmd: TCL AA: ++ Playing Welcome Prompt and options menu ++ Note: Martin Sloan has sent out detail about another BACD timer's behavior that is also contradict to the documentation in http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg33344.html Regards, --Somphol. ___ For more information regarding industry leading 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/B-ACD Documents in revamped Documentation Support page
Hi StefanoS, Just a tiny addition on the good collection of the link you've gathered, with the long navigation to Cisco IP Phone customization, you may want to get the the template from The url directories under enterprise parameters There you will see a URL that you can use to retrieve a template. If I am not mistaken, I think this tip is from one of William Bell's email or one of his blog posts. Regards, --Somphol On Thu, Oct 24, 2013 at 11:04 PM, StefanoS stefan...@gmail.com wrote: Oh never mind I found it! It looks like I'm too tired after all. So it's under Products Unified Communications Call Control Mid Market Call Control Cisco Unified Communication Manager Express Anyway since I've started this maybe we could make a new list with the new paths we might need in the lab exam. What do you think? Here are some more by me, which I'm sure you've already discovered yourselves, it's not space rocketry but it'll be a nice reference for the new ones.: CUPC Products Unified Communications Unified Communication Applications Messaging Cisco Unified Personal Communicator Release Notes 1-button Login (IPPA) Products Unified Communications Call Control Cisco Unified Communication Manager Configure Configuration Examples and TechNotes or Products Customer Collaboration Cisco Unified Contact Center Configure Configuration Examples and TechNotes Phone Customization Products Collaboration Endpoints IP Phones Cisco Unified IP Phone 7900 Series Maintain and Operate Maintain and Operate Guides IP Phone 7965G and 7945G Administration Guide for Cisco Unified Communications Manager 7.0 (SCCP and SIP) XML Customization for Phones Products Unified Communications Call Control Cisco Unified Communication Manager Configure Programming Guides Cisco Unified IP Phone Services Application Development Notes, Release 7.0(1) CiscoIPPhoneMenu CME Products Unified Communications Call Control Mid Market Call Control Cisco Unified Communication Manager Express On Thu, Oct 24, 2013 at 2:51 PM, StefanoS stefan...@gmail.com wrote: Hello everyone. This is a silly question, maybe I'm too tired but I'll ask anyway. A couple of days before Cisco did a rearrangement in Documentation Support page. So for example the section for UCM documents went under Products Unified Communications Call Control, or phones under Products Collaboration Endpoints Phones etc. I've found some but I can't find the path for the CME category and B-ACD docs path anywhere. It's not under CUCM (Call Manager) in Call Control section as I was expecting. So where is it? Thank's in advance. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading 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] Base configs and Dial Plan mapping
The one thing I'm really struggling with is mapping out my dial-plan during my read through of the lab. I would love to hear what others are doing. In my previous attempts, I find it very hard too, because the questions are verbose and I could either spend too much time reading OR not able to encode it into the table form correctly in haste OR simply skipped and waste too much time re-reading it. Here is my plan for my next attempt.I think the key is to have my pre-fabricate table then I will create my table quickly and ONLY adjust it while I read the question. So, I would quickly create this template. The list is there for easy cut paste. I will only complete Site A in during the lab, then I will just copy to SiteB SiteC. The rest is just modification of the table. (Note: I find that using TAB make it easier to align the columns, it could be 3 or 4 TABs.) In essence, focus on [1] Pre-fabrication [2] Quick to reproduce as a template. The rest is depending on how quick you can decipher verbose question and re-adjust the table. Get the screenshot here is the following format is bad - https://www.dropbox.com/s/o1ftbh4katnhjcu/Quick-Dial-Plan-Template.jpg, The TXT version is here - https://www.dropbox.com/s/m2p2lte5ezk6q7d/Quick-Dial-Plan-Template.TXT I think Matthew Berry youtube is good too - http://www.youtube.com/watch?v=4mP5powuFUM.I just think it can be complement by the cookie-cutting approach to encode the table in Notepad that can be reproduced quickly. === The LIST === ISDN Unknown Subscriber National International Any === SiteA === Calling Called Emer 7D / Unknown / ISDN 7D / Unknown / ISDN Local 7D / Subscriber / ISDN 7D / Subscriber / ISDN LD 10D / National / ISDN 10D / National / ISDN Intl 1+10D / International / ISDN 011! / International / ISDN Another variation of the table format is too cater for TEHO scenario or BACKUP Gateway scenario. === SiteA === Calling Called Emer 7D / Unknown / ISDN 7D / Unknown / ISDN Local 7D / Subscriber / ISDN 7D / Subscriber / ISDN Local(Backup via SiteB) 7D / Subscriber / ISDN 7D / Subscriber / ISDN LD 10D / National / ISDN 10D / National / ISDN Intl 1+10D / International / ISDN 011! / International / ISDN Regards, --Somphol. On Fri, Oct 18, 2013 at 12:22 PM, VanBenschoten, Brian brian.vanbenscho...@corebts.com wrote: I've found the QoS questions are very specific to test a certain area of knowledge. They are not looking for what we would consider a best practice system wide. I think we could skip setting the DSCP values in CUCM. If you think the question calls for it you can have your class-map match both AF31 and CS3 for signaling. *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Bill Hatcher *Sent:* Thursday, October 17, 2013 3:14 PM *To:* probert...@gmail.com *Cc:* ccievoice *Subject:* Re: [OSL | CCIE_Voice] Base configs and Dial Plan mapping I agree to setting the service parameters to default first. I was planning on doing that myself. As to changing the DSCP values, it all depends on what they ask for in the QoS section of the test is all. On Thu, Oct 17, 2013 at 2:24 PM, probert...@gmail.com probert...@gmail.com wrote: Hi, I think my strategy will be to set all Service Parameters to default before making changes. This way I can avoid and undesirable presets. Let me know your thoughts on this. Also why are you setting DSCP for Phone Configuration and DSCP for Cisco CallManager to Device Interface to AF31? Default CS3 should be good, let me know if I'm wrong on this? Rob On Thu, Oct 17, 2013 at 1:39 PM, Bill Hatcher wchatc...@gmail.com wrote: My test is just a couple of weeks away, and I've been reading different blogs on how to maximize your time. The one thing I'm really struggling with is mapping out my dial-plan during my read through of the lab. I would love to hear what others are doing. I have also been building base router configs for h323, gatekeeper, mgcp, srst,sip, etc so that I can practice quickly configuring those on the routers. One of the things I haven't really been keeping track of are some of the service parameters that I should adjust out of habit. Here are a few that I can think of off the top of my head that I plan on tweaking at the start of the exam. Please feel free to add to them. Enterprise Parameters DSCP for Phone Configuration - Set to AF31 DSCP for Cisco CallManager to Device Interface - Set to AF31 Change the Phone URL's to IP's Organization Top Level Domain Cluster Fully Qualified Domain Name Service Parameters - CallManager T302 Time - Know where it is if you need ot change interdigit timeout Call Classification - Offnet Builtin Bridge Enabled - True Device Name of
Re: [OSL | CCIE_Voice] Base configs and Dial Plan mapping
Hi Bill, I have adjusted it a little bit more to reduce the tab realignment while editing. It turns out that this creates more space where I can put in extra annotation such as route pattern. I think creating it from scratch could be done within 3 minutes. See a bit of a video clip here - http://www.youtube.com/watch?v=Cl9nANVgbms The text file can also available here - https://www.dropbox.com/s/t9bb2yo6x0wc66g/Dial-Plan-on-Notepad-Demo.TXT Regards, --Somphol. On Sat, Oct 19, 2013 at 1:15 AM, Bill Hatcher wchatc...@gmail.com wrote: Great template!! I like doing it this way better than on paper. I make to many mistakes on paper and can hardly read what I wrote. Thanks!! On Fri, Oct 18, 2013 at 2:53 AM, Somphol Boonjing somp...@gmail.comwrote: The one thing I'm really struggling with is mapping out my dial-plan during my read through of the lab. I would love to hear what others are doing. In my previous attempts, I find it very hard too, because the questions are verbose and I could either spend too much time reading OR not able to encode it into the table form correctly in haste OR simply skipped and waste too much time re-reading it. Here is my plan for my next attempt.I think the key is to have my pre-fabricate table then I will create my table quickly and ONLY adjust it while I read the question. So, I would quickly create this template. The list is there for easy cut paste. I will only complete Site A in during the lab, then I will just copy to SiteB SiteC. The rest is just modification of the table. (Note: I find that using TAB make it easier to align the columns, it could be 3 or 4 TABs.) In essence, focus on [1] Pre-fabrication [2] Quick to reproduce as a template. The rest is depending on how quick you can decipher verbose question and re-adjust the table. Get the screenshot here is the following format is bad - https://www.dropbox.com/s/o1ftbh4katnhjcu/Quick-Dial-Plan-Template.jpg, The TXT version is here - https://www.dropbox.com/s/m2p2lte5ezk6q7d/Quick-Dial-Plan-Template.TXT I think Matthew Berry youtube is good too - http://www.youtube.com/watch?v=4mP5powuFUM.I just think it can be complement by the cookie-cutting approach to encode the table in Notepad that can be reproduced quickly. === The LIST === ISDN Unknown Subscriber National International Any === SiteA === Calling Called Emer 7D / Unknown / ISDN 7D / Unknown / ISDN Local 7D / Subscriber / ISDN 7D / Subscriber / ISDN LD 10D / National / ISDN 10D / National / ISDN Intl 1+10D / International / ISDN 011! / International / ISDN Another variation of the table format is too cater for TEHO scenario or BACKUP Gateway scenario. === SiteA === Calling Called Emer 7D / Unknown / ISDN 7D / Unknown / ISDN Local 7D / Subscriber / ISDN 7D / Subscriber / ISDN Local(Backup via SiteB) 7D / Subscriber / ISDN 7D / Subscriber / ISDN LD 10D / National / ISDN 10D / National / ISDN Intl 1+10D / International / ISDN 011! / International / ISDN Regards, --Somphol. On Fri, Oct 18, 2013 at 12:22 PM, VanBenschoten, Brian brian.vanbenscho...@corebts.com wrote: I've found the QoS questions are very specific to test a certain area of knowledge. They are not looking for what we would consider a best practice system wide. I think we could skip setting the DSCP values in CUCM. If you think the question calls for it you can have your class-map match both AF31 and CS3 for signaling. *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Bill Hatcher *Sent:* Thursday, October 17, 2013 3:14 PM *To:* probert...@gmail.com *Cc:* ccievoice *Subject:* Re: [OSL | CCIE_Voice] Base configs and Dial Plan mapping I agree to setting the service parameters to default first. I was planning on doing that myself. As to changing the DSCP values, it all depends on what they ask for in the QoS section of the test is all. On Thu, Oct 17, 2013 at 2:24 PM, probert...@gmail.com probert...@gmail.com wrote: Hi, I think my strategy will be to set all Service Parameters to default before making changes. This way I can avoid and undesirable presets. Let me know your thoughts on this. Also why are you setting DSCP for Phone Configuration and DSCP for Cisco CallManager to Device Interface to AF31? Default CS3 should be good, let me know if I'm wrong on this? Rob On Thu, Oct 17, 2013 at 1:39 PM, Bill Hatcher wchatc...@gmail.com wrote: My test is just a couple of weeks away, and I've been reading different blogs on how to maximize your time. The one thing I'm really struggling with is mapping out my dial-plan during my read through of the lab. I would love to hear what others are doing. I have also been
Re: [OSL | CCIE_Voice] Base configs and Dial Plan mapping
+1 for that. Awesome information. Imagine this come out in one of the next revision of the exam on backup Route List and on purpose remove the ability to change this parameter on Call Manager. (4 points)!!! Thank you very much for sharing. Regards, --Somphol. On Sat, Oct 19, 2013 at 3:10 AM, Bill Hatcher wchatc...@gmail.com wrote: That's great information Bill. I think I might start leveraging that command on my real world deployments. Sent from my iPhone so please excuse any spelling mistakes. Bill Hatcher On Oct 18, 2013, at 9:42 AM, William Bell b...@ucguerrilla.com wrote: Bill, You can read about the command here: http://www.cisco.com/en/US/docs/ios/12_3t/voice/command/reference/vrht_d1_ps5207_TSD_Products_Command_Reference_Chapter.html#wp1459139 The important bit is: When the *dial-peer outbound status-check pots *command is configured, if the voice-port configured under an outbound POTS dial-peer is down, that dial-peer is excluded while matching the corresponding destination-pattern. Therefore, if there are no other matching outbound POTS dial-peers for the specified destination-pattern, the gateway will disconnect the call with a cause code of 1 (Unallocated/unassigned number), So, when you have this command enabled (default) AND you have a single PRI AND that PRI is down, call set up request from UCM to the VG will result in a response of unallocated/unassigned. Why? Because we have told the router to monitor the status of the PRI and intelligently detect when it is down. When it is down, the dial-peer is no longer evaluated during call setup. By turning this off, we are basically telling the VG to go ahead and try to use the busted PRI. Which then results in a different kind of setup error that will let the CUCM know it should continue hunting through its RG/RL configuration. Lots of people leverage the service parameter I mentioned below to route around PRIs that are off line. That is probably fine for the purposes of the IE lab. I prefer to disable status checking at the GW level. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Oct 18, 2013, at 10:06 AM, Bill Hatcher wrote: Bill, One other question, I'm not familiar with the command no dial-peer out status pots What's it do? On Fri, Oct 18, 2013 at 7:44 AM, William Bell b...@ucguerrilla.comwrote: I documented my strategy in my blog if interested. Part 2 in the series focuses on building various tables and the read-through portion of the exam: http://ucguerrilla.com/2013/04/ccie-voice-lab-strategy-part-2-read.html Looking back at my notes, I have the following Ent / Service params that I updated by default: *Enterprise Parameters:* * * - Auto Registration Protocol: SCCP - BLF for Call Lists: Enabled - Advertise G722 Codec: Disabled - URL Authentication: set IP instead of name - URL Directories: set IP instead of name - URL information: set IP instead of name - URL Services: set IP instead of name - Connection Monitor Duration: 60 (or do this at a device pool level) *Service Parameters* - BRQ Enabled: True - T302 timer: 5000 - H225 T302 timer: 5000 - G722 codec enabled: Disabled - iLBC codec enabled: Disabled - Intraregion Audio codec default: G729 - Inter-region Audio codec default: G729 - Automated Alternate Routing: True - Enable Mobile Voice Access: True - Inbound Calling Search Space for Remote Destination: Remote Destination Profile + Line Calling Search Space - System Remote Access Block Numbers: update as needed - Transfer on-hook enabled: True - Display Original Calling Number on Transfer from Unity: True - Max Forward unregistered hops to DN: 1 - Allow peer to preserve h323 calls: True/*need to add appropriate configuration on h323*/ Another service parameter I have seen people modify is the stop routing on unallocated number parameter. People mod this to allow calls to hunt around a H323 gateway that has a PRI which is down. I didn't use this method because I think it is the wrong approach to fixing that problem. I leveraged the IOS config command: no dial-peer out status pots HTH. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Oct 17, 2013, at 1:39 PM, Bill Hatcher wrote: My test is just a couple of weeks away, and I've been reading different blogs on how to maximize your time. The one thing I'm really struggling with is mapping out my dial-plan during my read through of the lab. I would love to hear what others are doing. I have also been building base router configs for h323, gatekeeper, mgcp, srst,sip, etc so that I can practice quickly configuring those on the routers. One of the things I haven't really been keeping track of are some of the service parameters that I should adjust out of habit. Here are a few that I can
Re: [OSL | CCIE_Voice] Base configs and Dial Plan mapping
Hi, On service parameters, you may also want to check Vik's article http://blog.ipexpert.com/2010/10/13/common-ucm-service-parameters-to-change/ . On the comment section, Trinifox also mentioned Please add: Intraregion Audio Codec Default to G729 to avoid CSCsl74701 Bug. In my checklist, I also tweak Conference section esp. Drop Ad Hoc Conference. Regards, --Somphol. On Fri, Oct 18, 2013 at 7:14 AM, Bill Hatcher wchatc...@gmail.com wrote: I agree to setting the service parameters to default first. I was planning on doing that myself. As to changing the DSCP values, it all depends on what they ask for in the QoS section of the test is all. On Thu, Oct 17, 2013 at 2:24 PM, probert...@gmail.com probert...@gmail.com wrote: Hi, I think my strategy will be to set all Service Parameters to default before making changes. This way I can avoid and undesirable presets. Let me know your thoughts on this. Also why are you setting DSCP for Phone Configuration and DSCP for Cisco CallManager to Device Interface to AF31? Default CS3 should be good, let me know if I'm wrong on this? Rob On Thu, Oct 17, 2013 at 1:39 PM, Bill Hatcher wchatc...@gmail.comwrote: My test is just a couple of weeks away, and I've been reading different blogs on how to maximize your time. The one thing I'm really struggling with is mapping out my dial-plan during my read through of the lab. I would love to hear what others are doing. I have also been building base router configs for h323, gatekeeper, mgcp, srst,sip, etc so that I can practice quickly configuring those on the routers. One of the things I haven't really been keeping track of are some of the service parameters that I should adjust out of habit. Here are a few that I can think of off the top of my head that I plan on tweaking at the start of the exam. Please feel free to add to them. Enterprise Parameters DSCP for Phone Configuration - Set to AF31 DSCP for Cisco CallManager to Device Interface - Set to AF31 Change the Phone URL's to IP's Organization Top Level Domain Cluster Fully Qualified Domain Name Service Parameters - CallManager T302 Time - Know where it is if you need ot change interdigit timeout Call Classification - Offnet Builtin Bridge Enabled - True Device Name of GK-controlled Trunk That Will Use Port 1720 - Set as needed. Transfer On-hook Enabled - True (Also a great thing to do in production when migrating users from other phone systems) Block offnet to offnet transfers - Know where it's at. Auto Call Pickup Enabled - True Call Back Enabled Flag - True (Verify) Single Button Barge/CBarge Policy - Set to Barge unless otherwise directed. Stop routing on Unallocated Number Flag - False - H323 redundancy Preferred G.711 Millisecond Packet Size - 20 (Verify) Preferred G.729 Millisecond Packet Size - 20 (Verify) G722 Codec Enabled - Disabled (Unless otherwise directed) Intraregion Audio Codec Default - G711/G722 (Verify) Interregion Audio Codec Default - G729 (Verify) Automated Alternate Routing Enabled - True (This one gets me every time on AAR so I turn it on by default now) Enable Mobile Voice Access - Set as required Mobile Voice Access Number - Set as required System Remote Access Blocked Numbers - Set as required Service Parameters -Cisco IP Voice Media Streaming App Supported MOH Codecs - G711 mulaw and G729 Annex A HTH Bill Hatcher ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension
Sorry for revisiting this old thread. The Calling Party Transformation at the Device Pool level would come in handy for this particular need. In the document starting 7.1.2, this is stated explicitly, http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_1_2/ccmfeat/fscallpn.html#wp1325305 . *Cisco Unity/Cisco Unity Connection* Because no calling party transformation options exist in Cisco Unified Communications Manager Administration for voice-messaging ports, make sure that you configure the calling party number transformations in the device pool that is associated with the voice-messaging ports. ... Table 7-8 Configuring the Calling Party Transformation CSS to Localize the Calling Party Number Also mentioned Use Device Pool Calling Party Transformation CSS as a method to Localize the Calling Party Number. ... ... The same document for 7.0.1 contained the table 7-8, but somehow doesn't have that explicit section on Cisco Unity/Cisco Unity Connection's calling party localization. (REF: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmfeat/fscallpn.html#wp1276877) So, I am not so sure whether this is possible in CUCM 7.0.1/CUC 7.0.1. I don't have lab access to test this now, but would appreciate if anyone can help testing this. Note: I recall seeing some sort of Technotes outlining the strategy to perform Calling Party transformation for Call Manager 4.x or something that doesn't rely on Gateway's Calling Party Transformation.I can't locate it now, but if anyone could point me to the URL that would be great. Regards, --Somphol. On Sat, Mar 23, 2013 at 12:34 AM, Leslie Meade leslie.me...@lvs1.comwrote: Easy way of doing this is to copy the hunt pilot and give it another number.. set user caller ID and mask it to Then in the call-manager-fallback change the voicemail to the new hunt pilot and your done ** ** ** ** *Leslie Meade* .. * Mobile:778.228.4339* | *Main:* *604.676.5239* *Email:* leslie.me...@lvs1.com ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Chadi H Hassoune (chassoun) *Sent:* Thursday, March 21, 2013 7:10 PM *To:* Pixar Perfect; Mark Thrash (marthras); Steve Keller *Cc:* CCIE Voice OSL *Subject:* Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension ** ** Calling Party Xform and assign it to the CUC Device Pool works fine for me. ** ** HTH ** ** *From: *Pixar Perfect pixarperf...@live.com *Date: *Wednesday, March 20, 2013 11:43 PM *To: *Mark Thrash (marthras) marth...@cisco.com, Steve Keller skeller...@gmail.com *Cc: *CCIE Voice OSL ccie_voice@onlinestudylist.com *Subject: *Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension ** ** the requirement is always for SiteB calling into SiteA voicemail by hitting Messages button. SiteA is always MGCP gateway. Calling Party Xform isnt any use on MGCP gateway for incoming calls. ** ** here is another way of doing it ... ** ** Voicemail Pilot for CUC is 2200 ** ** call-manager-fallback voicemail 2777 --- siteB specific ** ** translation-pattern on CUCM to convert 2777 into 2200 and mask calling number . The CSS of the translation pattern should have access to 2200. ** ** ** ** ** ** there is no definitive answer as to which solution is graded positively. there is a reason why many leading CCIE instructors say this is not a test of best practices but a test of how like able is your solution to the script. .. :) ** ** ** ** ** ** ** ** -- From: marth...@cisco.com To: skeller...@gmail.com Date: Thu, 21 Mar 2013 03:59:48 + CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension What about a calling party transform mask on the incoming gateway? Sent from my iPhone On Mar 20, 2013, at 10:43 PM, Steve Keller skeller...@gmail.com wrote: Thanks Bill, I like this option pretty well as it seems to limit treatment of calls this way to CUC when site B is in SRST mode only. I will try to lab this up tomorrow morning. Question for you, will this only solve my issue of pressing the VM button to access my mailbox to retrieve a message. Meaning when PSTN calls in to site B phone and then gets forward(redirected) to voicemail, I use a dial-peer that provides RDNIS capabilites to route the caller to the correct mailbox and not the opening greeting. So with this would i still want to use the following to get the caller into my mailbox? dial-peer voice 2600 pots destination-pattern 2600 port 0/0/0:23 no digit-strip prefix 202555 ( assuming no LD code at this site ) this is the way i get callers into my
[OSL | CCIE_Voice] CFUR vs Calling Party Transformation on MGCP gateway
Hi All, Am I understand correctly that unlike voice translation profile on IOS gateway, the calling party translation pattern, that is applied to gateway level for outgoing call, can't be tailored based on destination route pattern? For example, assuming both Site B and Site C are in SRST mode, for outgoing call originating to Site B and Site C number based on CFUR, there is no way to tailor the calling party number so that the calling number is 10D for call going to Site B and 11D for call going to Site C. If Site A gateway is H323 because these can be customized using voice translation profile, but this would be particularly impossible if Site A gateway is MGCP. Regards, --Somphol. ___ For more information regarding industry leading 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] CUPC port for QOS
Products - Voice and Unified Communications - Unified Communications Applications - Unified Communications Clients - Cisco Unified Personal Communicator Look at Release Notes http://www.cisco.com/en/US/docs/voice_ip_comm/cupc/7_1/english/release/notes/cupc71.html Regards, --Somphol. On Tue, Sep 3, 2013 at 3:56 PM, Karen Johnson karen.johnson...@yahoo.cawrote: folks, in exam,where to find CUPC port quickly to config the QOS lan ? K ___ For more information regarding industry leading 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] Generate a report for number of calls into PRI
Hi Hesham, On Wed, Sep 4, 2013 at 4:23 AM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: the number of concurrent calls on each PRI for example today from 8am to 5PM. I played around with SNMP to collect that values for a while. I remember that there is no MIBS OID for concurrent calls on MGCP's interface. You can achieve that via some sort of Perfmon AXL. The easiest seems to be via RTMT, but that can't be automated. Regards, --Somphol. ___ For more information regarding industry leading 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] Fwd: LAN QoS Basics
Hi Sam, I think I came across similar error message. You may want to adjust your Voice-Sig class-map as following: ! class-map match-any Voice-Sig match ip dscp cs3 af31 === list them in the same line class-map match-any Voice-RTP match ip dscp ef ! On Thu, Aug 29, 2013 at 4:47 PM, Sam Wilson wilsonc...@gmail.com wrote: ! class-map match-any Voice-Sig match ip dscp cs3 match ip dscp af31 class-map match-any Voice-RTP match ip dscp ef ! --Somphol. ___ For more information regarding industry leading 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] Frame-Relay Sh policy-map interface
You need to put in extra layer of policy-map for this. class-map match-any RTP match protocol rtp class-map match-any SIG match protocol skinny match protocol sip match protocol h323 match protocol custom-01 ! policy-map LLQ class SIG bandwidth 192 class RTP priority 64 ** ! policy-map FR-Branch1 class class-default shape average 364000 !-- not sure if this is mandatory service-policy LLQ map-class frame-relay FR-MC-BR1 service-policy output FR-Branch1 On Sun, Aug 18, 2013 at 5:09 PM, wilson.sam...@bt.com wrote: Hi All, ** ** For some reason I am not able to configure the Frame Policy-map applied via the map-class command: ** ** CorpHQ#sh policy CorpHQ#sh policy-map inter ** ** CorpHQ# ** ** I have checked everything and nothing seem to be wrong. ** ** Is there any special / hidden command that I shall need to invoke, in order to apply the Policy-Map on FR Subinterface via the map-class. ** ** Any help would be greatly appreciated. ** ** Please find below my configuration: ** ** ** ** Regardss ** ** ** ** ** ** ** ** class-map match-any RTP match protocol rtp class-map match-any SIG match protocol skinny match protocol sip match protocol h323 match protocol custom-01 ! policy-map FR-Branch1 class SIG bandwidth 192 class RTP priority 64 ** ** ** ** interface Serial0/0/1:0 description == Frame-Relay Circuit to WAN no ip address encapsulation frame-relay no keepalive cdp enable no frame-relay inverse-arp frame-relay lmi-type ansi ! interface Serial0/0/1:0.1 point-to-point description == FR To BR1 ip address 177.0.101.1 255.255.255.0 snmp trap link-status frame-relay interface-dlci 101 class FR-MC-BR1 ** ** ! map-class frame-relay FR-MC-BR1 service-policy output FR-Branch1 ** ** ** ** ** ** ** ** ** ** ** ** ** ** Kind Regards Wilson Samuel ** ** Wilson Samuel | East Region | BT Global Services |wilson.sam...@bt.com| http://globalservices.bt.com/ This e-mail contains information from BT which may be privileged or confidential. It's meant only for the individual(s) or entity named above. If you're not the intended recipient, note that disclosing, copying, distributing or using this information is prohibited. If you've received this e-mail in error, please let me know immediately on the e-mail address above. Thank you. We monitor our e-mail system, and may record your e-mails. ** ** BT Americas Inc., 150 Newport Avenue Ext., Quincy, MA 02171 USA BT Americas Inc. is a wholly owned subsidiary of British Telecommunications plc. ** ** ** ** Think before you print! Consider the environment before printing this email! ** ** ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Access list for cue traffic marking
The other point that also worth bearing in mind is this one concerning command mls qos, http://www.cisco.com/en/US/docs/switches/lan/catalyst3750/software/release/12.2_50_se/command/reference/cli1.html#wp5046030 . In essence, with 'mls qos' turned on, the default port trust state on all ports is untrusted. It is easy to forget this especially when you focus your attention to a single port on the switch under time limitation. mls qos ... ... Defaults QoS is disabled. There is no concept of trusted or untrusted ports because the packets are not modified (the CoS, DSCP, and IP precedence values in the packet are not changed). Traffic is switched in pass-through mode (packets are switched without any rewrites and classified as best effort without any policing). *When QoS is enabled with the mls qos global configuration command and all other QoS settings are set to their defaults, traffic is classified as best effort (the DSCP and CoS value is set to 0) without any policing. No policy maps are configured. The default port trust state on all ports is untrusted. The default ingress and egress queue settings are in effect.* Regards, --Somphol. On Mon, Aug 12, 2013 at 12:19 PM, Somphol Boonjing somp...@gmail.comwrote: This might be worth revisiting.Forgive me if this is not entirely a new insight. In short, be aware that as soon as the command service-policy input XXX in entered into the configuration, the mls qos trust cos/dscp will be removed. Likewise, if the command mls qos trust cos/dscp is re-entered, the command service-policy input XXX will be automatically removed. http://www.cisco.com/en/US/products/hw/switches/ps5023/products_tech_note09186a0080883f9e.shtml - If any other QoS Classification methods, such as port based or VLAN based, are configured on the port gi 1/0/3, those configurations are removed when you apply the policy-map. For example, the port Gi 1/0/13 is configured to trust CoS as shown here: interface GigabitEthernet1/0/13 description Access Port switchport access vlan 10 switchport mode access switchport voice vlan 100 mls qos cos 3 mls qos trust cos spanning-tree portfast - When you apply the policy-map to the interface, it removes the *trust* command. Distribution1(config)#*int gi 1/0/13* Distribution1(config-if)#*service-policy input sample-policy1* Distribution1(config-if)#*do show run int gi 1/0/13* Building configuration... Current configuration : 228 bytes ! interface GigabitEthernet1/0/13 description Access Port switchport access vlan 10 switchport mode access switchport voice vlan 100 service-policy input sample-policy1 *!--- It replaces the mls qos trust or mls qos !--- vlan-based command.* mls qos cos 3 *!--- This command is not removed.* spanning-tree portfast end Regards, --Somphol. On Mon, Jul 8, 2013 at 12:42 PM, jainpiyush2...@ymail.com wrote: Steve, you absolutely make sense that traffic for cue can be marked on router (site c) on which cue module is connected when it goes out on wan link.. and then on the trunk port on hq switch we would have trust statement. However the question in lab expect us to mark the cue traffic on hq switch on the port connected to sub cucm.. so the above solution won't work.. right? Thanks and regards, Piyush Jain Sent from my android device. -Original Message- From: sbar...@mystictraveler.net To: LorenzLGRC lorenzl...@gmail.com, Piyush Jain jainpiyush2...@ymail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Mon, 08 Jul 2013 6:23 AM Subject: RE: [OSL | CCIE_Voice] Access list for cue traffic marking Maybe I am missing something so please forgive me, and to recap, the question was LAN QoS and CUE (not WAN). The example below (which is pretty much out of the SRND) will correctly mark the traffic, but only going out the serial port. It seems to me that you would want to mark the traffic inbound from the CUE module to the router in which it resides so that no matter how the traffic exits the router it will be handled correctly. Can you mark the traffic as it leaves the AIM module and is passed to the router? As far as the policy map on the serial port, wouldn't we want to see all traffic correctly prioritized not just the CTI-QBE to answer the question correctly if we were to look at the WAN QoS? I assume for traffic leaving on an LAN port to a switch, the switch would have the appropriate trust statements and since we marked on the packets as they transition from the AIM to the router prioritization and re-marking would not be an issue? Steve Original Message Subject: Re: [OSL | CCIE_Voice] Access list for cue traffic marking From: LorenzLGRC lorenzl...@gmail.com Date: Sun, July 07, 2013 5:25 am To: Piyush Jain
Re: [OSL | CCIE_Voice] Per VC Frame Relay Fragmentation
I have rephrased my question slightly to highlight my dilemma which involves whether or not to configure fragmentation on all PVCs or only one out of three. *Given below detail:* HQ - SA - 64Kbps (DLCI 100) - Data Voice HQ - SB - 128Kbps (DLCI 200) - Data Only HQ - SC - 64Kbps (DLCI 300) - Data Only HQ - 256Kbps --- Aggregate WAN SA - 64Kbps SB - 128Kbps SC - 64Kbps 3xG729 between HQ-SA *Question: Configure FRF.12 with 10-ms delay for voice traffic.* [1] According to Table 3-1 Recommended Fragment Sizes, CIR, and Bc Values for Slow-Speed Frame Relay Links, it should be safe to use PVC speed as a reference point to calcualte Maximum Fragment Size (for 10-ms Delay). (As opposed to a physical interface's speed.) - http://www.cisco.com/en/US/docs/solutions/Enterprise/WAN_and_MAN/QoS_SRND/WANQoS.html#wp106984 [2] Should I perform the fragmentation on DLCI 200 DLCI 300 or not? I think it is reasonable to assume that since all of these PVCs will share the same physical interface, fragmenting only for large frame in DLCI 100 is not enough, therefore I think it is necessary to also fragment DLCI 200 DLCI 300. [For reference, under section FRF.12 on this link, it is stated - http://www.cisco.com/en/US/tech/tk652/tk698/technologies_configuration_example09186a0080094af9.shtml ...Any other PVCs that share the same physical interface need to configure the fragmentation to the size used by the voice PVC...] *If I have to bet, should I bet on performing fragmentation on all PVCs or only perform fragmentation on HQ-SA's PVC?* *Sample configuration below:* Class-map match-any signal match ip dscp cs3 Class-map match-any voice match ip dscp ef policy-map LLQ class voice priority 48 class signal bandwidth 8 class-default fair-queue policy-map SHAPE-SA class class-default shape average 64000 service-policy LLQ-SA policy-map SHAPE-SB class class-default shape average 128000 fair-queue policy-map SHAPE-SC class class-default shape average 64000 fair-queue map-class frame-relay HQ-SA frame-relay fragment 80 service-policy output SHAPE-SA map-class frame-relay HQ-SB frame-relay fragment 160 service-policy output SHAPE-SB map-class frame-relay HQ-SC frame-relay fragment 80 service-policy output SHAPE-SC interface serial 0/0 encapsulation frame-relay interface serial 0/0.1 point-to-point ip address 192.168.1.1 255.255.255.0 frame-relay interface-dlci 100 class HQ-SA interface serial 0/0.2 point-to-point ip address 192.168.2.1 255.255.255.0 frame-relay interface-dlci 200 class HQ-SB interface serial 0/0.3 point-to-point ip address 192.168.3.1 255.255.255.0 frame-relay interface-dlci 300 class HQ-SC Regards, --Somphol. On Tue, Aug 6, 2013 at 6:16 PM, Somphol Boonjing somp...@gmail.com wrote: Hi, Can anyone help confirm my understanding on this topic? My observation is that Per VC fragmentation, while it can be configured as when in the example below, is not very useful if not configured for all of the existing PVC that shared the same physical interface, isn't it? With the example below, only one of the VC (DLCI 100) is configured for fragmentation while the rest of the VCs (DLCI 200 DCLI 300) that shared the same physical interface are not, then potentially outgoing fragmented frames from DLCI 100 could be waiting in queue while a fragmented large data frames from DLCI 200/DLCI 300 is being sent out. Am I correct? (REF: http://www.cisco.com/en/US/docs/ios-xml/ios/wan_frly/configuration/12-4t/wan-mqc-fr-tfshp.html#GUID-BAC1F514-EBD4-48FF-87AB-41F2BF86463E ) Class-map voice match ip dscp ef policy-map llq class voice priority 32 policy-map shape-policy-map class class-default shape average 64000 shape adaptive 32000 service-policy llq map-class frame-relay shape-map-class frame-relay fragment 80 service-policy output shape-policy-map interface serial 0/0 encapsulation frame-relay interface serial 0/0.1 point-to-point ip address 192.168.1.1 255.255.255.0 frame-relay interface-dlci 100 class shape-map-class interface serial 0/0.2 point-to-point ip address 192.168.2.1 255.255.255.0 frame-relay interface-dlci 200 interface serial 0/0.3 point-to-point ip address 192.168.3.1 255.255.255.0 frame-relay interface-dlci 300 Regards, --Somphol ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Access list for cue traffic marking
This might be worth revisiting.Forgive me if this is not entirely a new insight. In short, be aware that as soon as the command service-policy input XXX in entered into the configuration, the mls qos trust cos/dscp will be removed. Likewise, if the command mls qos trust cos/dscp is re-entered, the command service-policy input XXX will be automatically removed. http://www.cisco.com/en/US/products/hw/switches/ps5023/products_tech_note09186a0080883f9e.shtml - If any other QoS Classification methods, such as port based or VLAN based, are configured on the port gi 1/0/3, those configurations are removed when you apply the policy-map. For example, the port Gi 1/0/13 is configured to trust CoS as shown here: interface GigabitEthernet1/0/13 description Access Port switchport access vlan 10 switchport mode access switchport voice vlan 100 mls qos cos 3 mls qos trust cos spanning-tree portfast - When you apply the policy-map to the interface, it removes the *trust* command. Distribution1(config)#*int gi 1/0/13* Distribution1(config-if)#*service-policy input sample-policy1* Distribution1(config-if)#*do show run int gi 1/0/13* Building configuration... Current configuration : 228 bytes ! interface GigabitEthernet1/0/13 description Access Port switchport access vlan 10 switchport mode access switchport voice vlan 100 service-policy input sample-policy1 *!--- It replaces the mls qos trust or mls qos !--- vlan-based command.* mls qos cos 3 *!--- This command is not removed.* spanning-tree portfast end Regards, --Somphol. On Mon, Jul 8, 2013 at 12:42 PM, jainpiyush2...@ymail.com wrote: Steve, you absolutely make sense that traffic for cue can be marked on router (site c) on which cue module is connected when it goes out on wan link.. and then on the trunk port on hq switch we would have trust statement. However the question in lab expect us to mark the cue traffic on hq switch on the port connected to sub cucm.. so the above solution won't work.. right? Thanks and regards, Piyush Jain Sent from my android device. -Original Message- From: sbar...@mystictraveler.net To: LorenzLGRC lorenzl...@gmail.com, Piyush Jain jainpiyush2...@ymail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Mon, 08 Jul 2013 6:23 AM Subject: RE: [OSL | CCIE_Voice] Access list for cue traffic marking Maybe I am missing something so please forgive me, and to recap, the question was LAN QoS and CUE (not WAN). The example below (which is pretty much out of the SRND) will correctly mark the traffic, but only going out the serial port. It seems to me that you would want to mark the traffic inbound from the CUE module to the router in which it resides so that no matter how the traffic exits the router it will be handled correctly. Can you mark the traffic as it leaves the AIM module and is passed to the router? As far as the policy map on the serial port, wouldn't we want to see all traffic correctly prioritized not just the CTI-QBE to answer the question correctly if we were to look at the WAN QoS? I assume for traffic leaving on an LAN port to a switch, the switch would have the appropriate trust statements and since we marked on the packets as they transition from the AIM to the router prioritization and re-marking would not be an issue? Steve Original Message Subject: Re: [OSL | CCIE_Voice] Access list for cue traffic marking From: LorenzLGRC lorenzl...@gmail.com Date: Sun, July 07, 2013 5:25 am To: Piyush Jain jainpiyush2...@ymail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Hello, you can use something like this: access-list 101 permit tcp host a.b.c.d any eq 2748 ! class-map match-all cti-qbe match access-group 101 ! policy-map cti-qbe class cti-qbe set dscp af31 bandwidth 20 ! interface Serial0/1 service-policy output cti-qbe On Sun, Jul 7, 2013 at 6:06 AM, Piyush Jain jainpiyush2...@ymail.comwrote: Hi Guys, I am trying to understand how we can mark CUE traffic on HQ Switch to implement LAN QOS. I have come up with the below solution. ip access-list extended name CUE permit tcp host 142.100.64.12 host 142.1.66.253 eq 2748 class-map match-any CUE-CLASS match access group name CUE policy-map CUE-POLICY class CUE-CLASS set ip dhcp CS3 int fa 1/0/4 description * CONNECTED TO SUB CUCM *** service policy input CUE-POLICY In above config, 142.100.64.12 is SUB CUCM, 142.1.66.253 is CUE on SC router. Explanation: Since we are applying service policy in incoming direction on switch port connected to CUCM, so the source port number (of CUCM) can be anything but destination port number (i.e for CUE) should be 2748 (JTAPI port). Any advice or inputs are most welcome. Cheers !! Piyush Jain
Re: [OSL | CCIE_Voice] mva
This question is definitely one of those, one that seems very simple either to confirm or deny. Either Yes it is 100% support or No it is 100% not. But googling around, and it is very confusing. Part of it may be because this problem only happens to in-call via Mobile Voice Access IVR. (The one you need to enter your PIN then press one to make a call.) Reading through two thread originated by a candidate by the name datucha/datoc, read through it both discussion threads, you will see how confusing this can be. http://ieoc.com/forums/p/18782/162049.aspx http://onlinestudylist.com/archives/ccie_voice/2012-February/079707.html From http://onlinestudylist.com/archives/ccie_voice/2012-February/079716.html, Vik's answer is below: *No- it's not supported. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vmalhi at ipexpert.com http://onlinestudylist.com/mailman/listinfo/ccie_voice* From http://ieoc.com/forums/p/18782/162049.aspx, Mark's answer is * Well, calling number shows up, only it will be what Cisco calls rooted in CDR as the shared desk line. Just as any call in to UCM where calling number matches the RD will show up as the shared desk line. * * Kind Regards,* * Mark Snow, CCIE #14073* Even Cisco document on this particular topic is very very slicky. *From CUCM 8.5.1 - http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_5_1/ccmfeat/fsmobmgr.html * * Caller ID—The system preserves and displays Caller ID on all calls. Users can take advantage of Mobile Connect with no loss of expected IP phone features.* *From CUCM 6.1.1 - http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/6_1_1/ccmfeat/fsmobmgr.html#wpxref99670 * * Caller ID—Caller ID is preserved and displayed on all calls. Users can take advantage of Mobile Connect with no loss of expected IP phone features. * Note that it says Mobile Connect and says nothing about Mobile Voice Access. Then you would assume it covers Mobile Voice Access too. Earlier definition of Mobile Voice Access in the same doc, also seems to suggest that MVA is indeed built on top of Mobile Connect, therefore reading in passing one would expect this to work. * Mobile Voice Access—This feature extends Mobile Connect capabilities by providing an interactive voice response (IVR) system to initiate two-stage dialed calls through the enterprise and activate or deactivate Mobile Connect capabilities. See the Mobile Voice Access sectionhttp://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_5_1/ccmfeat/fsmobmgr.html#wp1203108for a detailed discussion. * I think I will need to try to dig a bit deeper through Mobile Voice Access trace on CUCM to see what actually is happening.My scenario is simple, PSTN calls to MVA DID, Enter PIN, Press 1, then enter internal number with #.Even with this simplest case, involving no SLRG, the ***calling name (CNAM)* doesn't show up.Note also that my Remote Destination Number's Directory Number settings is setup correctly with *Display (Internal Caller ID) and **ASCII Display (Internal Caller ID)*. This values are confirmed to be working because when calling from the Mobile Connect (i.e. SNR) phone directly to one of the internal number, the number and name shows correctly. It would put this issue to a final rest too if we can find a relevenat Bug ID, if that is the case. Regards, --Somphol. On Sun, Aug 11, 2013 at 5:43 AM, Todd Carswell tcar0...@gmail.com wrote: The internal phone is known based on the remote destination configured for the user. To meet this requirement, the remote destination needs to be 10digits. For SNR calls outbound, strip off the area code, prepend a '9', and send to the appropriate gateway. Do not use SLRG. --Todd On Aug 10, 2013, at 11:51 AM, IE Target myfrnd...@gmail.com wrote: I am curious to know this mystery. As we are told that MVA caller should be able to call internal calls and it should appear as if it is coming form internal Phone. So if internal Phone calls it display caller id and calling name So MVA calls should also be displaying calling name. I heard it has also some thing to do with H.323 Display name, Fast/Slow start calls ? Any comments ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] mva
On Sun, Aug 11, 2013 at 11:17 AM, Somphol Boonjing somp...@gmail.comwrote: Reading through two thread originated by a candidate by the name datucha/datoc, read through it both discussion threads, you will see how confusing this can be. http://ieoc.com/forums/p/18782/162049.aspx http://onlinestudylist.com/archives/ccie_voice/2012-February/079707.html Having read through it one more time, I think there is no contradiction there. I would bet that Calling Name (CNAM) not showing up for call originating through MVA IVR is expected. Datoc said If anyone is interested: *Calling Number *is not supported for MVA calls into extenstions (or even any other destination). Got this answer from one of the CCIE Voice Instructors. Note that Datoc misspell that while he actually meant to say Calling Name. A typo which Mark's later on point out and hence his following comment. Well, calling number shows up, only it will be what Cisco calls rooted in CDR as the shared desk line. Just as any call in to UCM where calling number matches the RD will show up as the shared desk line. AND I think maybe you meant to say calling name (CNAM), not calling number (CLID) - that's what both Juan and Vik mentioned to you on OSL. To which Datoc later on agree. Yes i mean Calling Name :) Sorry for that, I just make a error in typing. So, while there is no Bug ID whatsoever, credible sources suggestion seems to be in line with real world experiment. I would be highly interested to see anyone who can produce different result for this paritcular case. (Note: Strictly for call made through MVA IVR.) Just an observation, there are a few points that worth discussing about, but this is getting long, so perhaps we can discuss it at a later point. Those topics are - MVA DID (appeared in CUCM's Call Manager service parameters) vs MVA DN (appeared on one of the Media Resource sub-menu) - The actual role of a Media Resource called MVA DN. Regards, --Somphol. ___ For more information regarding industry leading 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
On Fri, Aug 9, 2013 at 1:15 PM, Alex Mendoza aa.mend...@icloud.com wrote: Calling from my SNR to MVA is working, MVA asks for my pin number, then press 1, after that I dialed internal 4 digit extension but this internal phone only shows the caller number and not the caller name. I think is normal behavior, but when a calling from my SNR directly to a internal extension, it shows the caller number and the caller id. I am seeing the same thing for my MVA setup.I also presume this is expected behavior, but I'm not able to find any bug report or any concrete Cisco document to back it up though. Some people seems to report that the name will display after the call is connected. I can't reproduce that behavior, just the caller number for me during ringing and connected state of the call --- when made via MVA pilot number. Call directly from SNR number, i.e. Enterprise Feature Access, seems to be no problem with both Calling Name and Number. I'll be interested to see whether anyone else have different outcome. Regards, --Somphol. ___ For more information regarding industry leading 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] Per VC Frame Relay Fragmentation
Hi, Can anyone help confirm my understanding on this topic? My observation is that Per VC fragmentation, while it can be configured as when in the example below, is not very useful if not configured for all of the existing PVC that shared the same physical interface, isn't it? With the example below, only one of the VC (DLCI 100) is configured for fragmentation while the rest of the VCs (DLCI 200 DCLI 300) that shared the same physical interface are not, then potentially outgoing fragmented frames from DLCI 100 could be waiting in queue while a fragmented large data frames from DLCI 200/DLCI 300 is being sent out. Am I correct? (REF: http://www.cisco.com/en/US/docs/ios-xml/ios/wan_frly/configuration/12-4t/wan-mqc-fr-tfshp.html#GUID-BAC1F514-EBD4-48FF-87AB-41F2BF86463E ) Class-map voice match ip dscp ef policy-map llq class voice priority 32 policy-map shape-policy-map class class-default shape average 64000 shape adaptive 32000 service-policy llq map-class frame-relay shape-map-class frame-relay fragment 80 service-policy output shape-policy-map interface serial 0/0 encapsulation frame-relay interface serial 0/0.1 point-to-point ip address 192.168.1.1 255.255.255.0 frame-relay interface-dlci 100 class shape-map-class interface serial 0/0.2 point-to-point ip address 192.168.2.1 255.255.255.0 frame-relay interface-dlci 200 interface serial 0/0.3 point-to-point ip address 192.168.3.1 255.255.255.0 frame-relay interface-dlci 300 Regards, --Somphol ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Access list for cue traffic marking
Hi, Forgive me to chime in and most likely this won't answer the original question outright. Hesham's response is fairly inspiring in a way. It inspires me think a lot harder about this area. - Take another closer look at what is available, it is intriguing to see Layer 3 information such as IP Address header is now being inspected at a Layer 2 port through access-list and MQC. This is similar to the fact that Switch are now able to mare/re-mark DSCP which is at the IP header level. In all cases, it is likely to be a very simple IP header inspection and won't be able to do the full Layer 7 inspection. - Traffic classification when apply to Switch port, it will be subject to being [1] a very simply packet filtering tool such as access-list [2] different based on the direction.i.e. access-list to catch incoming traffic to device connecting to the switch vs the access-list to catch outgoing traffic from that device is going to be asymmetric. - If I would want to test the knowledge of traffic flow, while I can put the actual wording of the requirement differently, the core of the question would be Do you know all or some of the *incoming* traffic that flow from such as such to the rest of the infrastructure OR to a certain component of the infrastructure? With a simple twist, the question can be slightly changed to ask. Do you know all or some of the *outgoing* traffic that flow from such as such to the rest of the infrastructure OR to a certain component of the infrastructure? Now, replace such as such with: - CUCM Publisher - CUCM Subscriber - CUE - CUC Server - UCCX Server - CUPS Server - H323 Gateways - H323 Gatekeepers - IOS-based Media Resources - DHCP Server - TFTP Server - IP Phones - Attendant Console workstation - Agent Desktop workstation - CUPC Client Based on these basic ingredient, I guess someone can make a large set of exam bank out of it. This sort of question will be hard to crack if someone comes across a variation of it for the first time in the lab.It would be even harder to crack if someone try to memorize a set of access-list without really understand why it is the way it is. i.e. when to use access-list 101 tcp any any host x.x.x.x 2000 vs access-list 101 tcp host x.x.x.x 2000 any any. I don't dispute the probability of unreasonableness of the grading process itself, but this could also be a sound explanation of why it is very hard to crack. Regards, --Somphol. On Sun, Jul 7, 2013 at 8:38 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Guys , Get the biggest relief in your life. If you see this CUE QOS question just give it up. No one has ever scored that CUE Switch QOS and many people tried different things. My only advice give it up completely and never waste ur time or energy solving it. That particular lab is very long and if you have 2 hours left then try to play with it and enjoy. Knowing that the guys who passed this lab still didn't score that question in particular. In order for that question to be solved that needs to be consulted to a very knowledgable Routing and Switching . SP and Voice simultaneously even though the Cisco grading would be different than the real realistic world. To conclude , Never waste ur time or energy solve this stupid question trust me. Your passing score is 80% and this stupid question could be about 4% of the whole test. I know for fact that every minor mark counts in the total but its really up to the destiny. To me CCIE Test is no longer a test that you are real knowledgable or not. I definitely believe 100% CCIE test is like a gambling game , Jackpot or a roulette in LAS VEGAS. Don't have the faith that this thing is graded fairly with a standard. On 7 July 2013 02:25, LorenzLGRC lorenzl...@gmail.com wrote: Hello, you can use something like this: access-list 101 permit tcp host a.b.c.d any eq 2748 ! class-map match-all cti-qbe match access-group 101 ! policy-map cti-qbe class cti-qbe set dscp af31 bandwidth 20 ! interface Serial0/1 service-policy output cti-qbe On Sun, Jul 7, 2013 at 6:06 AM, Piyush Jain jainpiyush2...@ymail.comwrote: Hi Guys, I am trying to understand how we can mark CUE traffic on HQ Switch to implement LAN QOS. I have come up with the below solution. ip access-list extended name CUE permit tcp host 142.100.64.12 host 142.1.66.253 eq 2748 class-map match-any CUE-CLASS match access group name CUE policy-map CUE-POLICY class CUE-CLASS set ip dhcp CS3 int fa 1/0/4 description * CONNECTED TO SUB CUCM *** service policy input CUE-POLICY In above config, 142.100.64.12 is SUB CUCM, 142.1.66.253 is CUE on SC router. Explanation: Since we are applying service policy in incoming direction on switch port connected to CUCM, so the source port number (of CUCM) can be anything but destination port number (i.e for CUE) should be 2748 (JTAPI port). Any advice or inputs are most welcome.
Re: [OSL | CCIE_Voice] B-ACD Problem
Hi Hesham, My first guess would be an extra DTMF somehow. You application is running correctly, but I seems to think that it has received an unintentional digit. I would try to isolate the problem first by make a POTS dial-peer and test call from PSTN.If the problem still persists, you may want to post relevant dial-peer. Regards, --Somphol. On Tue, Jul 2, 2013 at 8:24 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Dear Experts, I have configured B-ACD. I have been configuring that everyday for months. Today is the first time. when i call the pilot number it says You have entered an invalid option , for sales press 1 for customer service press 2 for dialing by extension please press 3 What could be the problem? Thanks, Hesham ___ For more information regarding industry leading 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] B-ACD Problem
dial-peer voice 4001 pots service app-b-acd-aa incoming called-number *4008* ! I mean a test call from PSTN to x*4008*. --Somphol. On Tue, Jul 2, 2013 at 10:15 PM, Somphol Boonjing somp...@gmail.com wrote: Hi Hesham / Khaled, Fully agreed with that Kaled on that typo. Just a few more thought, there is also possibility that this is the actual audio of the file flash:en_bacd_welcome.au. You have entered an invalid option , for sales press 1 for customer service press 2 for dialing by extension please press 3 You can use debug voip application script to quickly see what audio files are played. Is it only en_bacd_welcome.au that is played or en_bacd_invalidoption.au is played first then followed by en_bacd_welcome.au.? Another quick isolation point is at POTS dial-peer, I think a quick change to number other than 4000 would help isolating the issue even further. My rational is to scope down the problematic area. dial-peer voice 4001 pots service app-b-acd-aa incoming called-number 4008 ! Then, make a test call from PSTN. Not that there is anything obvious, but isolation will make it easier to focus. Regards, --Somphol. On Tue, Jul 2, 2013 at 9:32 PM, khaled Saholy khaled_sah...@hotmail.comwrote: Hi Hesham, here are my comments: -I see under the application , no service app-b-acd-a , is this typo error? It shouldn't preceded with no. -If you're using drop through option , change the (1) param welcome-prompt _bacd_welcome.auparam drop-through-prompt _bacd_welcome.au (2) paramspace english index 1 from 1 to 0 -And under service app-b-acd , change param number-of-hunt-grps 2 from 2 to 1 Try these changes and let us know how it went with you. Regards. Khaled -- Date: Tue, 2 Jul 2013 04:21:09 -0700 Subject: Re: [OSL | CCIE_Voice] B-ACD Problem From: heshamcentr...@gmail.com To: khaled_sah...@hotmail.com CC: ccie_voice@onlinestudylist.com Hi Khaled , Here you are below application no service app-b-acd-aa param voice-mail 4220 paramspace english index 1 param max-time-call-retry 700 param service-name app-b-acd param number-of-hunt-grps 1 param drop-through-option 1 paramspace english language en param handoff-string app-b-acd-aa param max-time-vm-retry 2 paramspace english location flash: param aa-pilot 4000 param second-greeting-time 60 param welcome-prompt _bacd_welcome.au param call-retry-timer 15 ! service app-b-acd param queue-len 15 param aa-hunt1 4500 param number-of-hunt-grps 2 param queue-manager-debugs 1 ! global service alternate default ! ! dial-peer voice 4000 voip service app-b-acd-aa destination-pattern 4000 session target ipv4:142.102.66.254 incoming called-number 4000 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 4001 pots service app-b-acd-aa incoming called-number 4000 no ephone-hunt 10 longest-idle ephone-hunt 10 longest-idle pilot 4500 list 4101, 4102 timeout 10, 10 ! On 2 July 2013 04:06, khaled Saholy khaled_sah...@hotmail.com wrote: Hi Hesham, Can you post the config of B-ACD and ephone-hunt ? Also the output of show flash: | in au Regards. Khaled -- Date: Tue, 2 Jul 2013 03:24:10 -0700 From: heshamcentr...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] B-ACD Problem Dear Experts, I have configured B-ACD. I have been configuring that everyday for months. Today is the first time. when i call the pilot number it says You have entered an invalid option , for sales press 1 for customer service press 2 for dialing by extension please press 3 What could be the problem? Thanks, Hesham ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] L2 overhead for QoS
Hi Aman, In case this help, the topic seems to be discussed before. E.g. in the following thread. - http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg06632.html - http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg31952.html Regards, --Somphol. On Fri, Jun 28, 2013 at 10:10 AM, Kapuria, Aman aman.kapu...@team.telstra.com wrote: Anyone? ** ** *Aman * ** ** *From:* Kapuria, Aman *Sent:* Wednesday, 26 June 2013 3:31 PM *To:* ccie_voice@onlinestudylist.com *Subject:* L2 overhead for QoS ** ** Hi All, ** ** What L2 overhead do you use for frf.12 and MLP when you calculate number of calls over a link? Different providers have different approach around this. QoS SRND says “Frame Relay adds 4 bytes of Layer 2 overhead; Frame Relay with FRF.12 adds 8 bytes.” With frf.12 voice packets don’t get fragmented, so do you use 4 bytes for your calculation or 12 or some other number? For those who have done this in lab and got 100% for QoS, can they please advise what value they used? Thanks in advance Aman ** ** ** ** ** ** ** ** ___ For more information regarding industry leading 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] L2 overhead for QoS
Hi Aman, But voice packets font get fragmented when you use frf.12. So why would you use 8 and not 4? Just to try to dig up some relevant information. *[1] On whether voice packet get fragmented.* I agreed fully with you. If configured correctly, the voice packet should not be fragemented. Frame Relay Fragmentation for Voice http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a00801142de.shtml ...FRF.12 stipulates that, when fragmentation is on for a data-link connection identifier (DLCI), there is fragmentation of only data frames that exceed the specified fragmentation size. *This arrangement allows small VoIP packets, which are not fragmented due to the size*, to be interleaved as frames between large data packets that have been fragmented into smaller frames *[2] On frame-relay header different between fragmented frame and non-fragmented frame.* *Fragmented Frame* Address 2 bytes + UI 1 byte + NPID 1 byte + 2 bytes Fragmentation Header + DATA + FCS 2 bytes = *8 bytes* OK, I would understand if some will also count a flag which is 7E to delimits the beginning of the frame. Flag 1 byte + Address 2 bytes + UI 1 byte + NPID 1 byte + Fragmentation Header 2 bytes + DATA + FCS 2 bytes = *9** bytes (including 1 byte flag)* *Unfragmented Frame Source 1*: http://docwiki.cisco.com/wiki/Frame_Relay, Figure 5 Address 2 bytes + DATA + FCS 2 bytes = *4 bytes* Flag 1 byte + Address 2 bytes + DATA + FCS 2 bytes = *5 bytes (including 1 byte flag)* *Unfragmented Frame Source 2: IETF Frame-Relay header, Section 3, * http://tools.ietf.org/html/rfc1490 Address 2 bytes + UI 1 byte + NLPID 1 byte + DATA + FCS 2 bytes = *6 bytes*(excluding 1 byte flag) Flag 1 byte + Address 2 bytes + UI 1 byte + NLPID 1 byte + DATA + FCS 2 bytes = *7 bytes* (including 1 byte flag) What is NLPID? It contains values for many different protocols including IP, CLNP and IEEE Subnetwork Access Protocol (SNAP). So, it is likely to identify the encapsulated protocol, i.e. protocol code. UI = 0x03 always *Unfragmented Frame Source 3*: Cisco and RFC 1940 Encapsulation Figure 2-13, Cisco Frame Relay Solutions Guide by Jonathan Chin http://books.google.com.au/books?id=GPuhnmjxLuQCpg=PA34lpg=PA34dq=Cisco+Frame+Relay+frame+headersource=blots=diBrIR-p_ksig=CYVzWCoH-iTzxXvMosrJlLSAC3Ihl=ensa=Xei=K0PNUY_8OIvVkwXEoID4CQved=0CGsQ6AEwBw#v=onepageq=Cisco%20Frame%20Relay%20frame%20headerf=false Address 2 bytes + Protocol Type 2 bytes + DATA + FCS 2 bytes = *6 bytes* Flag 1 byte + Protocol Type 2 bytes + Address 2 bytes + DATA + FCS 2 bytes = *7 bytes* (including 1 byte flag) * * Reference Summary: - Page 9 of http://www.broadband-forum.org/technical/download/FRF.12/frf12.pdf - Figure: Five Fields Comprise the Frame Relay Frame, http://docwiki.cisco.com/wiki/Frame_Relay - Section 3: Frame Format - http://tools.ietf.org/html/rfc1490 - Cisco and RFC 1940 Encapsulation Figure 2-13, Cisco Frame Relay Solutions Guide by Jonathan Chin http://books.google.com.au/books?id=GPuhnmjxLuQCpg=PA34lpg=PA34dq=Cisco+Frame+Relay+frame+headersource=blots=diBrIR-p_ksig=CYVzWCoH-iTzxXvMosrJlLSAC3Ihl=ensa=Xei=K0PNUY_8OIvVkwXEoID4CQved=0CGsQ6AEwBw#v=onepageq=Cisco%20Frame%20Relay%20frame%20headerf=false *[3] My thought? (Well, I could be wrong, so use your discretion.)* For unfragmented frame if you believe in source #1 + QoS SRND + SRND 7.x, yes, I think you should go ahead and use 4 bytes (5 bytes if you think FCS deserves to be part of calculation).At least, we know why we choose these numbers. (Also uses 4 bytes Frame Relay Header is SRND 7.x Table 3-10 Bandwidth Consumption with Layer 2 Headers Included http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/netstruc.html#wp1044772 ) I believe 7 bytes is more accurate based on both source #2 source #3. Yes, I disagree with both QoS SRND SRND 7.x. There is also a big confusion, when it comes to FRF.12.And, I think that is why people are unsure of how Proctor going to grade the lab. First, within the QoS SRND itself. Table 1-3 Voice Bandwidth (Including Layer 2 Overhead) Second, in some other document such as Voice Over IP - Per Call Bandwidth Consumption, http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml If they do in fact agree that FRF.12's fragmentation header is not applicable for voice packet, they would not have a column for* Frame-Relay w/FRF.12*. So, it could well be the case that the proctor may use FRF.12 column if the requirement specifies FRF.12 for LFI. They may use 4 bytes and refers to SRND and make an assumption that this is a trick question for candidate. *[4] More likely way out?* Cisco exam designer has known to have already budgeted for 10% margin of error. And, they will most likely know that this particular topic is controversial. I think it is more than likely that either you use 4 bytes or 7 bytes or 8 bytes for your calculation, it will all be well within the acceptable margin of error
Re: [OSL | CCIE_Voice] L2 overhead for QoS
Just one more thing I forget to highlight. FRF.12 specification page 9 ( http://www.broadband-forum.org/technical/download/FRF.12/frf12.pdf), stated that NLPID field must be set to 0xB1 to signify that the frame contains a fragment. Then, it make more sense to assume that the NLPID field exists for unfragmented frame as well both to signify that the frame is unfragmented (because it will contain the value that is not 0xB1) and to use it as a Protocol Type field. That's also why I believe the overhead size of *7 bytes* is likely to be more accurate for both Unfragmented IETF and Cisco Frame Relay Header than 4 bytes. Regards, --Somphol. On Fri, Jun 28, 2013 at 6:39 PM, Somphol Boonjing somp...@gmail.com wrote: Hi Aman, But voice packets font get fragmented when you use frf.12. So why would you use 8 and not 4? Just to try to dig up some relevant information. *[1] On whether voice packet get fragmented.* I agreed fully with you. If configured correctly, the voice packet should not be fragemented. Frame Relay Fragmentation for Voice http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a00801142de.shtml ...FRF.12 stipulates that, when fragmentation is on for a data-link connection identifier (DLCI), there is fragmentation of only data frames that exceed the specified fragmentation size. *This arrangement allows small VoIP packets, which are not fragmented due to the size*, to be interleaved as frames between large data packets that have been fragmented into smaller frames *[2] On frame-relay header different between fragmented frame and non-fragmented frame.* *Fragmented Frame* Address 2 bytes + UI 1 byte + NPID 1 byte + 2 bytes Fragmentation Header + DATA + FCS 2 bytes = *8 bytes* OK, I would understand if some will also count a flag which is 7E to delimits the beginning of the frame. Flag 1 byte + Address 2 bytes + UI 1 byte + NPID 1 byte + Fragmentation Header 2 bytes + DATA + FCS 2 bytes = *9** bytes (including 1 byte flag) * *Unfragmented Frame Source 1*: http://docwiki.cisco.com/wiki/Frame_Relay, Figure 5 Address 2 bytes + DATA + FCS 2 bytes = *4 bytes* Flag 1 byte + Address 2 bytes + DATA + FCS 2 bytes = *5 bytes (including 1 byte flag)* *Unfragmented Frame Source 2: IETF Frame-Relay header, Section 3, * http://tools.ietf.org/html/rfc1490 Address 2 bytes + UI 1 byte + NLPID 1 byte + DATA + FCS 2 bytes = *6 bytes * (excluding 1 byte flag) Flag 1 byte + Address 2 bytes + UI 1 byte + NLPID 1 byte + DATA + FCS 2 bytes = *7 bytes* (including 1 byte flag) What is NLPID? It contains values for many different protocols including IP, CLNP and IEEE Subnetwork Access Protocol (SNAP). So, it is likely to identify the encapsulated protocol, i.e. protocol code. UI = 0x03 always *Unfragmented Frame Source 3*: Cisco and RFC 1940 Encapsulation Figure 2-13, Cisco Frame Relay Solutions Guide by Jonathan Chin http://books.google.com.au/books?id=GPuhnmjxLuQCpg=PA34lpg=PA34dq=Cisco+Frame+Relay+frame+headersource=blots=diBrIR-p_ksig=CYVzWCoH-iTzxXvMosrJlLSAC3Ihl=ensa=Xei=K0PNUY_8OIvVkwXEoID4CQved=0CGsQ6AEwBw#v=onepageq=Cisco%20Frame%20Relay%20frame%20headerf=false Address 2 bytes + Protocol Type 2 bytes + DATA + FCS 2 bytes = *6 bytes* Flag 1 byte + Protocol Type 2 bytes + Address 2 bytes + DATA + FCS 2 bytes = *7 bytes* (including 1 byte flag) * * Reference Summary: - Page 9 of http://www.broadband-forum.org/technical/download/FRF.12/frf12.pdf - Figure: Five Fields Comprise the Frame Relay Frame, http://docwiki.cisco.com/wiki/Frame_Relay - Section 3: Frame Format - http://tools.ietf.org/html/rfc1490 - Cisco and RFC 1940 Encapsulation Figure 2-13, Cisco Frame Relay Solutions Guide by Jonathan Chin http://books.google.com.au/books?id=GPuhnmjxLuQCpg=PA34lpg=PA34dq=Cisco+Frame+Relay+frame+headersource=blots=diBrIR-p_ksig=CYVzWCoH-iTzxXvMosrJlLSAC3Ihl=ensa=Xei=K0PNUY_8OIvVkwXEoID4CQved=0CGsQ6AEwBw#v=onepageq=Cisco%20Frame%20Relay%20frame%20headerf=false *[3] My thought? (Well, I could be wrong, so use your discretion.)* For unfragmented frame if you believe in source #1 + QoS SRND + SRND 7.x, yes, I think you should go ahead and use 4 bytes (5 bytes if you think FCS deserves to be part of calculation).At least, we know why we choose these numbers. (Also uses 4 bytes Frame Relay Header is SRND 7.x Table 3-10 Bandwidth Consumption with Layer 2 Headers Included http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/netstruc.html#wp1044772 ) I believe 7 bytes is more accurate based on both source #2 source #3. Yes, I disagree with both QoS SRND SRND 7.x. There is also a big confusion, when it comes to FRF.12.And, I think that is why people are unsure of how Proctor going to grade the lab. First, within the QoS SRND itself. Table 1-3 Voice Bandwidth (Including Layer 2 Overhead) Second, in some other document such as Voice Over IP - Per Call Bandwidth Consumption, http
Re: [OSL | CCIE_Voice] Codec and CAC section
This may have nothing to do with the implementation checklist, but may be useful as part of a verification steps. Assuming SiteA SiteC are configured with a typical scenario with G711 inter-region and G729 intra-region, and RSVP is required. 1. Incoming or outgoing PSTN call to or from say SC Phone 1. Verify the codec of the active call using ?? that it is G711.Put the call on Hold. Assuming the MoH is in SiteA. Is it successful? Does music going through to the PSTN phone successfully? 2. Incoming or outgoing PSTN call to or from say SC Phone 1. Verify the codec of the active call using ?? that it is G711. Transfer the call to SA Phone 1. Is it successful? If yes, then from SA Phone 1, put the call on Hold.Is it successful? 3. Make a call from SC Phone 1 SA Phone 1, verify on both phone that the codec for the active call is G729.Verify on the gateway using the following commands to spot anything obviously problem, such as 80K is reserved instead of 24K, or G729abr8 or G729ar8 is used instead of G729, etc. # show ip rsvp interface # show ip rsvp installed # show sccp connections # show sccp connections rsvp # show sccp connections detail 4. (optional) Set CFWDALL on SC Ph1 to SA Ph1 DN. Make PSTN call to SC Ph1 DN, the call should ring SA Ph1.Answer the call, verify the active codec, etc. This is just my idea of a partial verification steps that may help isolate any problem in your configuration. Another thing is that I find useful for my study. I often create the following MTP for each site, put them in different MRG so that I can add/remove/re-order them in MRGL to see how CUCM select different resources. In case you find it useful too. - MTP: G711 only + RSVP - MTP: G729 only + RSVP - MTP: Pass-through only + RSVP - XCODER: XCODER + RSVP Hope this doesn't deviate too much from your question. Regards, --Somphol On Mon, Jun 24, 2013 at 4:26 AM, Karen Johnson karen.johnson...@yahoo.cawrote: hi folks, can anyone share experience on what to check on this section , I got 0 for few attempt. Here is what I did : UCM = - service parameter : no G722 and ILBC - Enterprise parameter G711 intra, G729 inter - Region : HQ SB SC, HQ-HQ : G711 , SB-SB G711, SC-SC : g711 (rest relation : G729) and assign tp DP - Location : HQ and SC : mandatory , assign to DP - MRGL HQ -- MRG-- MTP from HQ same for SC , assign to DP router HQ and SC = - dspfarm profile 3 mtp codec pass-through codec g729r8 rsvp maximum sessions software 4 (as they asked 4 session of g729) associate application SCCP - interface Serial0/0/0.1 point-to-point frame-relay interface-dlci 102 ip rsvp bandwidth 112 verification = - call hq to hq, sb sb : g711, inter site phone and GW : g729 - sh ip rsvp reservation : 40 k (ring) , and 24 k (connect) question: - did i miss something critical that cause the mark to be 0 ? ___ For more information regarding industry leading 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] Codec and CAC section
and it is not ideal as Media Resources in the same MRG is subject to be picked randomly, so I rely on MRGL for the order. What I find out so far is this. - MTP that are capable of doing both PASSTHROUGH codec and RSVP (i.e. PASSTHROUGH+RSVP MTP are only selected when there is no other RSVP-capable resource available. (i.e. internal calls between regions) - The problem with this type of MTP is that, it is dumb. It can't perform several smart MTP function such as mid-call supplementary service e.g. MoH, Call Transfer, etc. - With PASSTHROUGH+RSVP MTP, you won't be able to make PSTN call via H323 gateway and later on either try to Transfer it to the phone in other region or even to put it on hold. - You will find that both G711+RSVP MTP and XCODER+RSVP MTP are both involved to allow a PSTN call to be transfer successfully to a phone in other region. - For internal calls, I suspect that the order of preference for CUCM are XCODER+RSVP MTP, G729+RSVP MTP, PASSTHROUGH+RSVP. - PSTN call either incoming or outgoing to the SC Phone via SC Gateway will always involve an G711 MTP (assuming intra-region codec is G711).With that therefore if a call is needed to either be transfered to other region or to listen to G729 MoH, the Media Resource that is also needed is XCODER+RSVP MTP. - I haven't tried this, but I strongly suspect that if I enable G722 and set inter-region to G711, then PASSTHROUGH+RSVP MTP will be used even if all other types of MTP are available. (Those are just my observations and it could be wrong, so please verify in your lab.) I remember one case when I know that RSVP bandwidth is enough for 4 calls to go through, but I can only make three calls.I found out later on that because I am not careful with my MRG grouping, XCODER+RSVP is always used up first unnecessarily (i.e. even without any need for XCODER, just RSVP is needed).And I happen to limit it to 3 sessions.So, when I actually need it for one of my calls, i.e. a call that actually requires both XCODER+RSVP MTP, they are all used up. So, after all, what seems to be very straightforward, has a lot of possibilities to go wrong. I encourage you to lab it up if you think it makes sense. You may like what you learn in the process. Regards, --Somphol. On Tue, Jun 25, 2013 at 2:29 AM, Karen Johnson karen.johnson...@yahoo.cawrote: hi Somphol, thanks for your advice here. could you pls help me to undertsand the concern when we check this ? - Assuming the MoH is in SiteA. Is it successful? Does music going through to the PSTN phone successfully? ( do Cisco always expect G711 when they did not say in exam ? ) - Transfer the call to SA Phone 1. Is it successful? If yes, then from SA Phone 1, put the call on Hold.Is it successful? (do you mean to chekc if MOH g729 here?) - (optional) Set CFWDALL on SC Ph1 to SA Ph1 DN. Make PSTN call to SC Ph1 DN, the call should ring SA Ph1.Answer the call, verify the active codec, etc. (this should G729,right ?) - - MTP: G711 only + RSVP - MTP: G729 only + RSVP - MTP: Pass-through only + RSVP - XCODER: XCODER + RSVP (may i know what is purpose) - if they ask codec G711, we should see 80k insh rsvp reservation ? tks K *From:* Somphol Boonjing somp...@gmail.com *To:* Karen Johnson karen.johnson...@yahoo.ca *Cc:* ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com *Sent:* Monday, June 24, 2013 5:42:26 AM *Subject:* Re: [OSL | CCIE_Voice] Codec and CAC section This may have nothing to do with the implementation checklist, but may be useful as part of a verification steps. Assuming SiteA SiteC are configured with a typical scenario with G711 inter-region and G729 intra-region, and RSVP is required. 1. Incoming or outgoing PSTN call to or from say SC Phone 1. Verify the codec of the active call using ?? that it is G711.Put the call on Hold. Assuming the MoH is in SiteA. Is it successful? Does music going through to the PSTN phone successfully? 2. Incoming or outgoing PSTN call to or from say SC Phone 1. Verify the codec of the active call using ?? that it is G711. Transfer the call to SA Phone 1. Is it successful? If yes, then from SA Phone 1, put the call on Hold.Is it successful? 3. Make a call from SC Phone 1 SA Phone 1, verify on both phone that the codec for the active call is G729.Verify on the gateway using the following commands to spot anything obviously problem, such as 80K is reserved instead of 24K, or G729abr8 or G729ar8 is used instead of G729, etc. # show ip rsvp interface # show ip rsvp installed # show sccp connections # show sccp connections rsvp # show sccp connections detail 4. (optional) Set CFWDALL on SC Ph1 to SA Ph1 DN. Make PSTN call to SC Ph1 DN, the call should ring SA Ph1.Answer the call, verify the active codec, etc. This is just my idea of a partial verification steps that may help isolate any problem in your configuration
Re: [OSL | CCIE_Voice] Gatekeeper is unable to reach SiteB under Call Forward UnRegister
Hi Hesham, knowing that Gatekeeper is working with SiteB under normal operation but doesn't work with CFUR Could you please clarify the problem you are facing? What do you mean when you say the gatekeeper is not working with CFUR? Any Ideas, I think we will need to simplify the scenario to the level that we can understand the expected call flow correctly, then from there we can isolate problematic area further. 'debug isdn q931' on HQ GW and SiteB GW might also give us some more idea. Regards, --Somphol On Sun, Jun 23, 2013 at 12:45 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Dear Experts, SiteC is CME and connected with HQ and SB via Gatekeeper Gatekeeper is working excellent with HQ and SB I am configuring Call Forward Unregister for SiteB. SiteB has Call-Manager-Fallback mode working excellent Now, I have configured Call Forward Unregister in the service parameter I changed maximum hops to DN unregister is 1 I have Created a Partitions and CSS for CFUR I forward SiteB1 and SiteB2 telephones in unregisted internal and external to be 9723033001 with forward css CFUR-CSS I created Route List to point to HQ Router and create route pattern for CFUR Now gatekeeper is reaching both HQ and SiteB in normal operaiton when I put SiteB under call-manager-fallback mode when I dial from HQ 3001 the CFUR works and shows the E164 number when I dial from SiteC 3001 via gatekeeper it shows unknown number knowing that Gatekeeper is working with SiteB under normal operation but doesn't work with CFUR Any Ideas, Thanks, Hesham ___ For more information regarding industry leading 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] Gatekeeper is unable to reach SiteB under Call Forward UnRegister
Hi Hesham, Thanks for the detail explanation and well thanks for sharing the case. I find it very intriguing. I'm working on some idea, but for now, I just want to forward your reply to the group, in case anyone else can help too. --Somphol On Sun, Jun 23, 2013 at 4:44 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Hi Somphol, I have to give you details as much as I can for better assistance not to tackle some of the information. Ok let me tell you the call flow In my scenario HQ and SB are registered to CUCM and SC is a CME (SC is connected with HQ SB via Gatekeeper) I want to make sure that in case of SB WAN Failure HQ/SC phones are able to call Siteb phone using 4 digits in the event of wan failue.When you call from HQ phone calls should be routed through HQ gateway. When you call from SC Phones calls should be routed through the GK and then HQ Gateway. In normal operation the call flow is HQ dials 4xxx --- Gatekeeeper --- SC CME SB dials 4xxx --- Gatekeeper --- SC CME now when you configure Call Forward Unregister internal HQ dials 3XXX -- SB phone is no longer registered to CUCM and is configured for internal and external if Unregistered to be forwarded to 9723033001 Number is dialed on HQ Gateway by CFUR --- Call reaches SB via HQ PSTN Gateway successfully the Requirement now SC CME dials 3XXX---Call Router via Gatekeeper-- SB phone is no longer registered to CUCM and is configured for internal and external if Unregistered to be forwarded to 9723033001--- Number is dialed on HQ Gateway by CFUR --- Call reaches SB via HQ PSTN Gateway successfully. Now the current situation when SC CME dials 3XXX when the SB is under WAN Failure it goes no where after the Gatekeeper but when I switch back the SB Phones to be registered to CUCM rather than CALL MANAGER FALLBACK the call go through via Gatekeeper Many Thanks, Hesham On 22 June 2013 23:26, Somphol Boonjing somp...@gmail.com wrote: Hi Hesham, knowing that Gatekeeper is working with SiteB under normal operation but doesn't work with CFUR Could you please clarify the problem you are facing? What do you mean when you say the gatekeeper is not working with CFUR? Any Ideas, I think we will need to simplify the scenario to the level that we can understand the expected call flow correctly, then from there we can isolate problematic area further. 'debug isdn q931' on HQ GW and SiteB GW might also give us some more idea. Regards, --Somphol On Sun, Jun 23, 2013 at 12:45 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Dear Experts, SiteC is CME and connected with HQ and SB via Gatekeeper Gatekeeper is working excellent with HQ and SB I am configuring Call Forward Unregister for SiteB. SiteB has Call-Manager-Fallback mode working excellent Now, I have configured Call Forward Unregister in the service parameter I changed maximum hops to DN unregister is 1 I have Created a Partitions and CSS for CFUR I forward SiteB1 and SiteB2 telephones in unregisted internal and external to be 9723033001 with forward css CFUR-CSS I created Route List to point to HQ Router and create route pattern for CFUR Now gatekeeper is reaching both HQ and SiteB in normal operaiton when I put SiteB under call-manager-fallback mode when I dial from HQ 3001 the CFUR works and shows the E164 number when I dial from SiteC 3001 via gatekeeper it shows unknown number knowing that Gatekeeper is working with SiteB under normal operation but doesn't work with CFUR Any Ideas, Thanks, Hesham ___ For more information regarding industry leading 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] B-ACD
Remove all of the param under the service did the trick, Say you have this in your running config application service app-b-acd param number-of-hunt-grps 2 param aa-hunt1 param aa-hunt2 1222 param queue-len 15 param queue-manager-debugs 1 ! Then, application service app-b-acd no param number-of-hunt-grps 2 no param aa-hunt1 no param aa-hunt2 1222 no param queue-len 15 no param queue-manager-debugs 1 Once there is no param set for the service, it will be removed from the running-config. --- Detail trace below: --- Branch2#show run | begin application application service app-b-acd param queue-len 15 param aa-hunt1 param queue-manager-debugs 1 param aa-hunt2 1222 param number-of-hunt-grps 2 ! ! Branch2(config)#application Branch2(config-app)# service app-b-acd Branch2(config-app-param)#no param queue-len 15 Warning: parameter queue-len has not been registered under app-b-acd namespace Branch2(config-app-param)#no param aa-hunt1 Warning: parameter aa-hunt1 has not been registered under app-b-acd namespace Branch2(config-app-param)# Branch2(config-app-param)#do show run | begin application application service app-b-acd param queue-manager-debugs 1 param aa-hunt2 1222 param number-of-hunt-grps 2 ! ! Branch2(config-app-param)#no param queue-manager-debugs 1 Warning: parameter queue-manager-debugs has not been registered under app-b-acd namespace Branch2(config-app-param)#no param aa-hunt2 1222 Warning: parameter aa-hunt2 has not been registered under app-b-acd namespace Branch2(config-app-param)#no param number-of-hunt-grps 2 Warning: parameter number-of-hunt-grps has not been registered under app-b-acd namespace Branch2(config-app-param)#do show run | begin application associate application SCCP ! dspfarm profile 5 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 On Sun, Jun 23, 2013 at 12:20 AM, Bill Lake whl...@gmail.com wrote: Try doing all command not just these Sent from my iPhone On Jun 22, 2013, at 6:51 AM, CISCO CCIE VOICE ccievoic...@gmail.com wrote: Thanks Bill for your reply, I have done no service app-b-acd and no service app-b-acd-aa but showing all those commands in Running configuration thanks On Sat, Jun 22, 2013 at 1:48 PM, Bill Lake whl...@gmail.com wrote: If it is showing up in the running configuration, then you most likely see something like below, the best way to remove this is to no the commands Or to have done a Archive or copy of the config before you apply it. then restore that config as the startup and reboot. application * service app-b-acd * param number-of-hunt-grps 2 param aa-hunt2 param aa-hunt3 1222 param queue-len 15 param queue-manager-debugs 1 ! * service app-b-acd-aa * paramspace english index 1 paramspace english language en paramspace english location flash: param service-name app-b-acd param handoff-string app-b-acd-aa param aa-pilot 8005550123 param welcome-prompt _bacd_welcome.au param number-of-hunt-grps 2 param dial-by-extension-option 1 param second-greeting-time 60 param call-retry-timer 15 param max-time-call-retry 700 param max-time-vm-retry 2 param voice-mail 5003 ! dial-peer voice 222 voip service app-b-acd-aa destination-pattern 8005550123 session target ipv4:192.168.1.1 incoming called-number 8005550123 dtmf-relay h245-alphanumeric codec g711ulaw no vad On Sat, Jun 22, 2013 at 2:25 AM, Somphol Boonjing somp...@gmail.comwrote: That one is the embedded one so you actually can not remove it. However, you can simply ignore it and use one that is external script. So, if you have the external BACD script, you can use it instead of the embedded one. Branch2#show flash | inc bacd 107 30421bacd/app-b-acd-3.0.0.2.tcl 108 55599bacd/app-b-acd-aa-3.0.0.2.tcl application service *funnyqueue flash:/bacd/*app-b-acd-3.0.0.2.tcl -- you can you whatever name you like, in this case funnyqueue -- point the script to the script with correct path . (detail remove for brevity)... ! service* funnyaa flash:/bacd/app-b-acd-aa-3.0.0.2.tcl * -- you can you whatever name you like, in this case funnyaa -- point the script to the script with correct path . (detail remove for brevity). param service-name *funnyqueue* -- refer to your queue application name param handoff-string *funnyaa* . (detail remove for brevity). ! dial-peer voice 222 voip service *funnyaa* -- refer to your AA application name. . (detail remove for brevity)... ! To remove it from the running config, then you can, application no service *funnyqueue flash:/bacd/*app-b-acd-3.0.0.2.tcl no service* funnyaa flash:/bacd/app-b-acd-aa-3.0.0.2.tcl
Re: [OSL | CCIE_Voice] Gatekeeper is unable to reach SiteB under Call Forward UnRegister
Hi Hesham, I have a few ideas. I want to remove a few things out of the equation, first try to set codec for all inter-region to G711. Second, if you are using Local Route Group (LRG), replace it with a more straightforward settings -- i.e. point the RL directly to HQ gateway in your case for relevant route pattern. We can deal with them later on once we understand this case to the bone. There are two call legs. The first call leg is from SC PH1 to reach x3001 via a H323 Trunk on CUCM -- the Trunk with gatekeeper control. The call should be directed to the gatekeeper who in turn should be routing it to the H323 Trunk on CUCM. The H323 Trunk should have significant digits set to 4 and a CSS that can reach x3001. Upon hitting x3001, CUCM will discover that the number is forwarded to 9723033001. Assuming that you have set the CSS for CFUR on x3001 correctly, that will match a Router Pattern that route the call toward HQ Gateway.This is a second call leg.(If you use the LRG, at this point, the LRG for the incoming H323 Trunk will cause the call to route to the wrong RG.) Once the second call leg is established, then CUCM will tell the two parities to open the RTP channel directly to each other (i.e. between the CME and the HQ Gateway.) (Well, sort of, if you have MTP required check on the H323 Trunk, then an MTP will be involved.) You problem could be on either one of this. While I believe that since you can make a call from HQ PH1 to x3001 successfully, the problem may not be in the 2nd leg, I don't entirely want to rule out the CSS, the Significant digits as well as the fact that HQ PH1 and the incoming H323 Trunk will be more than likely belong to a different Device Pool Region. I think debug gatekeeper main 10 on the gatekeeper would help. On the H323 CUCM Trunk, RTMT Real Time monitoring with Detailed Debug turn on would help you see whether the H323 Trunk has the right CSS to reach x3001. Hope this gives you some idea to work on this case. Regards, --Somphol. On Sun, Jun 23, 2013 at 5:27 PM, Somphol Boonjing somp...@gmail.com wrote: Hi Hesham, Thanks for the detail explanation and well thanks for sharing the case. I find it very intriguing. I'm working on some idea, but for now, I just want to forward your reply to the group, in case anyone else can help too. --Somphol On Sun, Jun 23, 2013 at 4:44 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Hi Somphol, I have to give you details as much as I can for better assistance not to tackle some of the information. Ok let me tell you the call flow In my scenario HQ and SB are registered to CUCM and SC is a CME (SC is connected with HQ SB via Gatekeeper) I want to make sure that in case of SB WAN Failure HQ/SC phones are able to call Siteb phone using 4 digits in the event of wan failue.When you call from HQ phone calls should be routed through HQ gateway. When you call from SC Phones calls should be routed through the GK and then HQ Gateway. In normal operation the call flow is HQ dials 4xxx --- Gatekeeeper --- SC CME SB dials 4xxx --- Gatekeeper --- SC CME now when you configure Call Forward Unregister internal HQ dials 3XXX -- SB phone is no longer registered to CUCM and is configured for internal and external if Unregistered to be forwarded to 9723033001 Number is dialed on HQ Gateway by CFUR --- Call reaches SB via HQ PSTN Gateway successfully the Requirement now SC CME dials 3XXX---Call Router via Gatekeeper-- SB phone is no longer registered to CUCM and is configured for internal and external if Unregistered to be forwarded to 9723033001--- Number is dialed on HQ Gateway by CFUR --- Call reaches SB via HQ PSTN Gateway successfully. Now the current situation when SC CME dials 3XXX when the SB is under WAN Failure it goes no where after the Gatekeeper but when I switch back the SB Phones to be registered to CUCM rather than CALL MANAGER FALLBACK the call go through via Gatekeeper Many Thanks, Hesham On 22 June 2013 23:26, Somphol Boonjing somp...@gmail.com wrote: Hi Hesham, knowing that Gatekeeper is working with SiteB under normal operation but doesn't work with CFUR Could you please clarify the problem you are facing? What do you mean when you say the gatekeeper is not working with CFUR? Any Ideas, I think we will need to simplify the scenario to the level that we can understand the expected call flow correctly, then from there we can isolate problematic area further. 'debug isdn q931' on HQ GW and SiteB GW might also give us some more idea. Regards, --Somphol On Sun, Jun 23, 2013 at 12:45 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Dear Experts, SiteC is CME and connected with HQ and SB via Gatekeeper Gatekeeper is working excellent with HQ and SB I am configuring Call Forward Unregister for SiteB. SiteB has Call-Manager-Fallback mode working excellent Now, I have configured Call Forward
Re: [OSL | CCIE_Voice] Gatekeeper is unable to reach SiteB under Call Forward UnRegister
Hi Hesham, If the problem is on the gatekeeper, it could be as simple as the zone prefix not configured to point to CUCM for the pattern 3... Given that in normal situation, the zone prefix would be pointing SBGW either dynamically or statically. The configure with static zone prefix set would look similar to this. gatekeeeper ... ... gw-type-prefix 1#* default-technology zone prefix THEZONE 3... gw-priority 100 SBGW zone prefix THEZONE 3... gw-priority 10 CUCMTRUNK ... ... If your CUCM SBGW happens to be in the different zones, that is a different matter. Looking at a configuration guide for zone prefix command, I don't think it is possible for a zone prefix to point to two different local zones. (See: http://www.cisco.com/en/US/docs/ios/12_3/vvf_r/vrg_z1_ps1839_TSD_Products_Command_Reference_Chapter.html#wp1002271 ) So, in essence, I doubt that this would work. gatekeeeper ... ... gw-type-prefix 1#* default-technology zone prefix SBZONE 3... gw-priority 100 SBGW zone prefix CUCMZONE 3... gw-priority 10 CUCMTRUNK ... ... Regards, --Somphol. On Sun, Jun 23, 2013 at 6:45 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Hi Somphol, Of course all your sequence of ideas definitely make sense. However, I did exactly all that I made the Route List for CFUR is very specific to HQ Gateway and not SLRG. and Tried to change the Inbound Calls in the trunk and changed the CSS to INTERNAL and still didn't work, yes I am looking into the debug command that will show me the gatekeeper call flow. I have been a long time never worked with that. Thanks for your ideas, I will keep you and the forum posted if I got any updates, Thanks, Hesham On 23 June 2013 01:40, Somphol Boonjing somp...@gmail.com wrote: Hi Hesham, I have a few ideas. I want to remove a few things out of the equation, first try to set codec for all inter-region to G711. Second, if you are using Local Route Group (LRG), replace it with a more straightforward settings -- i.e. point the RL directly to HQ gateway in your case for relevant route pattern. We can deal with them later on once we understand this case to the bone. There are two call legs. The first call leg is from SC PH1 to reach x3001 via a H323 Trunk on CUCM -- the Trunk with gatekeeper control. The call should be directed to the gatekeeper who in turn should be routing it to the H323 Trunk on CUCM. The H323 Trunk should have significant digits set to 4 and a CSS that can reach x3001. Upon hitting x3001, CUCM will discover that the number is forwarded to 9723033001. Assuming that you have set the CSS for CFUR on x3001 correctly, that will match a Router Pattern that route the call toward HQ Gateway.This is a second call leg.(If you use the LRG, at this point, the LRG for the incoming H323 Trunk will cause the call to route to the wrong RG.) Once the second call leg is established, then CUCM will tell the two parities to open the RTP channel directly to each other (i.e. between the CME and the HQ Gateway.) (Well, sort of, if you have MTP required check on the H323 Trunk, then an MTP will be involved.) You problem could be on either one of this. While I believe that since you can make a call from HQ PH1 to x3001 successfully, the problem may not be in the 2nd leg, I don't entirely want to rule out the CSS, the Significant digits as well as the fact that HQ PH1 and the incoming H323 Trunk will be more than likely belong to a different Device Pool Region. I think debug gatekeeper main 10 on the gatekeeper would help. On the H323 CUCM Trunk, RTMT Real Time monitoring with Detailed Debug turn on would help you see whether the H323 Trunk has the right CSS to reach x3001. Hope this gives you some idea to work on this case. Regards, --Somphol. On Sun, Jun 23, 2013 at 5:27 PM, Somphol Boonjing somp...@gmail.comwrote: Hi Hesham, Thanks for the detail explanation and well thanks for sharing the case. I find it very intriguing. I'm working on some idea, but for now, I just want to forward your reply to the group, in case anyone else can help too. --Somphol On Sun, Jun 23, 2013 at 4:44 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Hi Somphol, I have to give you details as much as I can for better assistance not to tackle some of the information. Ok let me tell you the call flow In my scenario HQ and SB are registered to CUCM and SC is a CME (SC is connected with HQ SB via Gatekeeper) I want to make sure that in case of SB WAN Failure HQ/SC phones are able to call Siteb phone using 4 digits in the event of wan failue.When you call from HQ phone calls should be routed through HQ gateway. When you call from SC Phones calls should be routed through the GK and then HQ Gateway. In normal operation the call flow is HQ dials 4xxx --- Gatekeeeper --- SC CME SB dials 4xxx --- Gatekeeper --- SC CME now when you configure Call Forward Unregister internal HQ
Re: [OSL | CCIE_Voice] Gatekeeper is unable to reach SiteB under Call Forward UnRegister
Hi Hesham, Essentially, the gw-priority is to advise the gatekeeper to choose SBGW over CUCMTRUNK. The higher the number, the higher the priority. Without this it will distribute the call to 3XXX to both CUCMTRUNK and SBGW in a round robin fashion. If you give higher priority to SBGW, then call will be routed to SBGW unless it is not available. gw-type-prefix 1#* default-technology zone prefix THEZONE 3... gw-priority 100 SBGW zone prefix THEZONE 3... gw-priority 10 CUCMTRUNK I'm fairly new to gatekeeper myself, so it would be great if you can lab it up and see if I am wildly off the mark. Regards, --Somphol. On Sun, Jun 23, 2013 at 8:37 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Hi Somphol, HQ SB are in the same zone and i don't understand zone prefix THEZONE 3... gw-priority 100 SBGW I think I should disregard it as they are int he same zone It's all just the CUCM Trunk and has both 2XXX and 3XXX I think that could make it work Thank you very much for ur great input I will test it and let u know Thank you very much for ur great efforts. On Jun 23, 2013, at 3:30 AM, Somphol Boonjing somp...@gmail.com wrote: Hi Hesham, If the problem is on the gatekeeper, it could be as simple as the zone prefix not configured to point to CUCM for the pattern 3... Given that in normal situation, the zone prefix would be pointing SBGW either dynamically or statically. The configure with static zone prefix set would look similar to this. gatekeeeper ... ... gw-type-prefix 1#* default-technology zone prefix THEZONE 3... gw-priority 100 SBGW zone prefix THEZONE 3... gw-priority 10 CUCMTRUNK ... ... If your CUCM SBGW happens to be in the different zones, that is a different matter. Looking at a configuration guide for zone prefix command, I don't think it is possible for a zone prefix to point to two different local zones. (See: http://www.cisco.com/en/US/docs/ios/12_3/vvf_r/vrg_z1_ps1839_TSD_Products_Command_Reference_Chapter.html#wp1002271 ) So, in essence, I doubt that this would work. gatekeeeper ... ... gw-type-prefix 1#* default-technology zone prefix SBZONE 3... gw-priority 100 SBGW zone prefix CUCMZONE 3... gw-priority 10 CUCMTRUNK ... ... Regards, --Somphol. On Sun, Jun 23, 2013 at 6:45 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Hi Somphol, Of course all your sequence of ideas definitely make sense. However, I did exactly all that I made the Route List for CFUR is very specific to HQ Gateway and not SLRG. and Tried to change the Inbound Calls in the trunk and changed the CSS to INTERNAL and still didn't work, yes I am looking into the debug command that will show me the gatekeeper call flow. I have been a long time never worked with that. Thanks for your ideas, I will keep you and the forum posted if I got any updates, Thanks, Hesham On 23 June 2013 01:40, Somphol Boonjing somp...@gmail.com wrote: Hi Hesham, I have a few ideas. I want to remove a few things out of the equation, first try to set codec for all inter-region to G711. Second, if you are using Local Route Group (LRG), replace it with a more straightforward settings -- i.e. point the RL directly to HQ gateway in your case for relevant route pattern. We can deal with them later on once we understand this case to the bone. There are two call legs. The first call leg is from SC PH1 to reach x3001 via a H323 Trunk on CUCM -- the Trunk with gatekeeper control. The call should be directed to the gatekeeper who in turn should be routing it to the H323 Trunk on CUCM. The H323 Trunk should have significant digits set to 4 and a CSS that can reach x3001. Upon hitting x3001, CUCM will discover that the number is forwarded to 9723033001. Assuming that you have set the CSS for CFUR on x3001 correctly, that will match a Router Pattern that route the call toward HQ Gateway.This is a second call leg.(If you use the LRG, at this point, the LRG for the incoming H323 Trunk will cause the call to route to the wrong RG.) Once the second call leg is established, then CUCM will tell the two parities to open the RTP channel directly to each other (i.e. between the CME and the HQ Gateway.) (Well, sort of, if you have MTP required check on the H323 Trunk, then an MTP will be involved.) You problem could be on either one of this. While I believe that since you can make a call from HQ PH1 to x3001 successfully, the problem may not be in the 2nd leg, I don't entirely want to rule out the CSS, the Significant digits as well as the fact that HQ PH1 and the incoming H323 Trunk will be more than likely belong to a different Device Pool Region. I think debug gatekeeper main 10 on the gatekeeper would help. On the H323 CUCM Trunk, RTMT Real Time monitoring with Detailed Debug turn on would help you see whether the H323 Trunk has the right CSS to reach x3001. Hope this gives you some idea to work
Re: [OSL | CCIE_Voice] Gatekeeper is unable to reach SiteB under Call Forward UnRegister
Sorry, I assume wrongly that SBGW will ever take the call for 3 Your normal path is for both 2... and 3... to be pointing to CUCMTRUNK only. Given that both SBGW and CUCMTRUNK are registered to the same zone, it would be necessary to exclude SBGW from ever getting the call destined to 2... or 3 gw-type-prefix 1#* default-technology zone prefix THEZONE 3... gw-priority 0 SBGW zone prefix THEZONE 3... gw-priority 10 CUCMTRUNK zone prefix THEZONE 2... gw-priority 0 SBGW zone prefix THEZONE 2... gw-priority 10 CUCMTRUNK Sorry for the confusion. Even if you don't have gw-priority, when SBGW is unreachable, it should not cause the problem and call should be sent correctly to CUCMTRUNK. Then, it is less likely that the problem would be in the gatekeeper call leg, unless you use some sort of tech-prefix in addition to zone prefix. Regards, --Somphol On Sun, Jun 23, 2013 at 8:43 PM, Somphol Boonjing somp...@gmail.com wrote: Hi Hesham, Essentially, the gw-priority is to advise the gatekeeper to choose SBGW over CUCMTRUNK. The higher the number, the higher the priority. Without this it will distribute the call to 3XXX to both CUCMTRUNK and SBGW in a round robin fashion. If you give higher priority to SBGW, then call will be routed to SBGW unless it is not available. gw-type-prefix 1#* default-technology zone prefix THEZONE 3... gw-priority 100 SBGW zone prefix THEZONE 3... gw-priority 10 CUCMTRUNK I'm fairly new to gatekeeper myself, so it would be great if you can lab it up and see if I am wildly off the mark. Regards, --Somphol. On Sun, Jun 23, 2013 at 8:37 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Hi Somphol, HQ SB are in the same zone and i don't understand zone prefix THEZONE 3... gw-priority 100 SBGW I think I should disregard it as they are int he same zone It's all just the CUCM Trunk and has both 2XXX and 3XXX I think that could make it work Thank you very much for ur great input I will test it and let u know Thank you very much for ur great efforts. On Jun 23, 2013, at 3:30 AM, Somphol Boonjing somp...@gmail.com wrote: Hi Hesham, If the problem is on the gatekeeper, it could be as simple as the zone prefix not configured to point to CUCM for the pattern 3... Given that in normal situation, the zone prefix would be pointing SBGW either dynamically or statically. The configure with static zone prefix set would look similar to this. gatekeeeper ... ... gw-type-prefix 1#* default-technology zone prefix THEZONE 3... gw-priority 100 SBGW zone prefix THEZONE 3... gw-priority 10 CUCMTRUNK ... ... If your CUCM SBGW happens to be in the different zones, that is a different matter. Looking at a configuration guide for zone prefix command, I don't think it is possible for a zone prefix to point to two different local zones. (See: http://www.cisco.com/en/US/docs/ios/12_3/vvf_r/vrg_z1_ps1839_TSD_Products_Command_Reference_Chapter.html#wp1002271 ) So, in essence, I doubt that this would work. gatekeeeper ... ... gw-type-prefix 1#* default-technology zone prefix SBZONE 3... gw-priority 100 SBGW zone prefix CUCMZONE 3... gw-priority 10 CUCMTRUNK ... ... Regards, --Somphol. On Sun, Jun 23, 2013 at 6:45 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Hi Somphol, Of course all your sequence of ideas definitely make sense. However, I did exactly all that I made the Route List for CFUR is very specific to HQ Gateway and not SLRG. and Tried to change the Inbound Calls in the trunk and changed the CSS to INTERNAL and still didn't work, yes I am looking into the debug command that will show me the gatekeeper call flow. I have been a long time never worked with that. Thanks for your ideas, I will keep you and the forum posted if I got any updates, Thanks, Hesham On 23 June 2013 01:40, Somphol Boonjing somp...@gmail.com wrote: Hi Hesham, I have a few ideas. I want to remove a few things out of the equation, first try to set codec for all inter-region to G711. Second, if you are using Local Route Group (LRG), replace it with a more straightforward settings -- i.e. point the RL directly to HQ gateway in your case for relevant route pattern. We can deal with them later on once we understand this case to the bone. There are two call legs. The first call leg is from SC PH1 to reach x3001 via a H323 Trunk on CUCM -- the Trunk with gatekeeper control. The call should be directed to the gatekeeper who in turn should be routing it to the H323 Trunk on CUCM. The H323 Trunk should have significant digits set to 4 and a CSS that can reach x3001. Upon hitting x3001, CUCM will discover that the number is forwarded to 9723033001. Assuming that you have set the CSS for CFUR on x3001 correctly, that will match a Router Pattern that route the call toward HQ Gateway.This is a second call leg.(If you use the LRG, at this point, the LRG for the incoming
Re: [OSL | CCIE_Voice] B-ACD
That one is the embedded one so you actually can not remove it. However, you can simply ignore it and use one that is external script. So, if you have the external BACD script, you can use it instead of the embedded one. Branch2#show flash | inc bacd 107 30421bacd/app-b-acd-3.0.0.2.tcl 108 55599bacd/app-b-acd-aa-3.0.0.2.tcl application service *funnyqueue flash:/bacd/*app-b-acd-3.0.0.2.tcl -- you can you whatever name you like, in this case funnyqueue -- point the script to the script with correct path . (detail remove for brevity)... ! service* funnyaa flash:/bacd/app-b-acd-aa-3.0.0.2.tcl * -- you can you whatever name you like, in this case funnyaa -- point the script to the script with correct path . (detail remove for brevity). param service-name *funnyqueue* -- refer to your queue application name param handoff-string *funnyaa* . (detail remove for brevity). ! dial-peer voice 222 voip service *funnyaa* -- refer to your AA application name. . (detail remove for brevity)... ! To remove it from the running config, then you can, application no service *funnyqueue flash:/bacd/*app-b-acd-3.0.0.2.tcl no service* funnyaa flash:/bacd/app-b-acd-aa-3.0.0.2.tcl* Ref: http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html#wp1026305 Compared to using the embedded one below: application service app-b-acd -- you can't change the name of the embedded BACD Queue script . (detail remove for brevity)... ! service app-b-acd-aa -- you can't change the name of the embedded BACD AA script . (detail remove for brevity). param service-name app-b-acd -- refer to the embedded BACD Queue script param handoff-string app-b-acd-aa . (detail remove for brevity). ! dial-peer voice 222 voip service app-b-acd-aa -- refer to the name of the embedded BACD AA script . (detail remove for brevity)... ! Ref: Embedded Call-Queue and AA Tcl Scripts: Example http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html On 22/06/2013 4:18 PM, CISCO CCIE VOICE ccievoic...@gmail.com wrote: application no service app-b-acd no service app-b-acd-aa On Sat, Jun 22, 2013 at 9:06 AM, Somphol Boonjing somp...@gmail.comwrote: Hi, Are you able to show part of the configuration that you have tried to remove from the running configuration? --Somphol On Sat, Jun 22, 2013 at 1:00 AM, CISCO CCIE VOICE ccievoic...@gmail.comwrote: Hi, I am trying to Remove B-ACD configuration but still showing in the running configuration i have restarted the router but no look any guess? thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] SRST checklist
To add to the list: - music-on-hold matching that of CUCM if applicable (i.e. in a situation where the file is provided) - secondary-dialtone - Alert Name Connected Name (i.e. ephone-dn's name parameter) (esp. intra-site) - CFUR settings for relevant DN on CUCM (To what extent this is relevant to SRST point, I don't know, but I think I will need to clarify this with the proctor if it is not clearly stated.) I've also tested that the following is not possible in SRST as the phone will still try to retrieve the file from the original TFTP server. - (not possible) matching Ringlist.xml / DistinctiveRingList.xml to that of CUCM --Somphol On Sat, Jun 22, 2013 at 9:20 AM, Ovidiu Popa ovi.p...@gmail.com wrote: Follow-up: voice service voip h323 call preserve Ovidiu On Fri, Jun 21, 2013 at 10:40 PM, Ovidiu Popa ovi.p...@gmail.com wrote: Adding one more for H323 gateways: *CUCM service Parameter : Allow Peer to Preserve H.323 Calls* * * Keep it coming guys and gals Regards, Ovidiu On Fri, Jun 21, 2013 at 11:55 AM, Somphol Boonjing somp...@gmail.comwrote: To add to your list that is already good, - date/time format - timezone - system message - number of max calls busy call triggers - call pickup behavior if applicable (directed vs no directed call pickup) - call-transfer pattern - call-forward pattern - number of channels for ephone (dual, octal) - cbarge for shared line if applicable - SRST for media resources (via sdspfarm) - cptone if applicable --Somphol. --Somphol On Fri, Jun 21, 2013 at 3:08 PM, Karen Johnson karen.johnson...@yahoo.ca wrote: all, i am trying to compile SRST check for my next attempt. I never got full mark here in my few attempts and always curious what i missing (even it seems I already done what they asked) - caller id and name ( hide or display) - Mwi light and VM message from PSTN and IP phone - inter site call VM inter site - COR if any - When forward call come , it play personal greeting - DND to divert - huntstop channel - if agents still working - after back to normal mode, verify everhthing -softkeys - feature : conference - timeout interdigit and Cfwd timer similar to UCM mode - always use preference 9 and dial-peer hunt 2 Any other tips and trick that I am not aware ? help please K - ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] UCM DHCP
A few points that I think worth double check: [1] Assuming this is the configuration of the DHCP on CUCM, I think the primary router doesn't require subnet mask to be specified DHCP Server : 10.10.210.10 subnet IPV4 address: 10.10.200.0 primary start addr: 10.10.200.120 primary end addr : 10.10.200.130 primary router : 10.10.200.3/255.255.255.0 [2] In the same broadcast domain (VLAN 102), you seem to have two routing interface, one on the router and another on as SVI on the switch.Unless it serves other purpose, I think you can safely remove the SVI. interface FastEthernet0/0.102 encapsulation dot1Q 102 ip address 10.10.200.3 255.255.255.0 ip helper-address 10.10.210.10 ! interface Vlan102 ip address 10.10.200.120 255.255.255.0 ! [3] Once the SVI interface vlan102 is removed, then the interface that actually forward your request to the CUCM DHCP should be the fa0/0.102 on the router that has already been configured with ip helper-address 10.10.210.10 (assuming that this is your CUCM IP). Then, the following trace on the switch will not be relevant anymore. 01:58:00: DHCPD: Finding a relay for client 0100.1bd4.c698.6c on interface Vlan102. 01:58:00: DHCPD: there is no pool for 10.10.200.120. Once removing SVI 102, then you should try to focus your debugging effort on the Router and not on the switch. [4] Your DHCP range seems to start with 10.10.200.120 and that could cause a collision with the SVI's IP address, even if everything else is correct. I remember that IOS-based DHCP seems to assign the last IP address in the pool, but I'm not sure how CUCM DHCP select the IP address from the pool. --Somphol On Sat, Jun 22, 2013 at 10:27 AM, anuritha konjety anurith...@gmail.comwrote: Ovidiu, Yes, ip helper-address is configured. Regards, Anu On Fri, Jun 21, 2013 at 4:32 PM, Ovidiu Popa ovi.p...@gmail.com wrote: Hello Anuritha Do you by any chance have an SVI on vlan 102 ? If yes have you tried configuring ip helper-address on that SVI ? Regards, Ovidiu On Sat, Jun 22, 2013 at 12:26 AM, anuritha konjety anurith...@gmail.comwrote: Hello, I am having trouble with getting phones Ip address when UCM(pub) is configured as the DHCP server. Following is partial config from the switch the router. I have made sure - 1. Dot1q trunk on HQ-RTR interface (on the switch). Make sure that all appropriate vlans are allowed on the trunk link and that native VLAN lines up (1 is default). 2. Ensure VLANs are provisioned correctly, assigned to the right interfaces, and active (sh vlan b) 3. Double check scope config on CUCM Pub. Check each parameter. 4. Made sure helper-address is configured 5. disabled CSA service from pub 6. restarted DHCP service several times DHCP Server : 10.10.210.10 subnet IPV4 address: 10.10.200.0 primary start addr: 10.10.200.120 primary end addr : 10.10.200.130 primary router : 10.10.200.3/255.255.255.0 debug ip dhcp server events: SiteA-Switch# $ 01:58:00: DHCPD: Reload workspace interface Vlan102 tableid 0. 01:58:00: DHCPD: tableid for 10.10.200.120 on Vlan102 is 0 01:58:00: DHCPD: client's VPN is . 01:58:00: DHCPD: Finding a relay for client 0100.1bd4.c698.6c on interface Vlan102. 01:58:00: DHCPD: there is no pool for 10.10.200.120. SiteA-Switch#$ 01:58:34: DHCPD: checking for expired leases. SiteA-Switch#$ 01:59:25: DHCPD: Reload workspace interface Vlan102 tableid 0. 01:59:25: DHCPD: tableid for 10.10.200.120 on Vlan102 is 0 01:59:25: DHCPD: client's VPN is . 01:59:25: DHCPD: Finding a relay for client 0100.1bd4.c698.6c on interface Vlan102. 01:59:25: DHCPD: there is no pool for 10.10.200.120. HQ Switch:- SiteA-Switch#sh run interface FastEthernet1/0/1 description TRUNK to HQ-RTR switchport trunk encapsulation dot1q switchport mode trunk speed 100 duplex full ! interface FastEthernet1/0/2 description HQ PHONE 1- 7960 phone switchport access vlan 101 switchport mode access switchport voice vlan 102 spanning-tree portfast ! interface FastEthernet1/0/3 ! --More-- interface FastEthernet1/0/4 description SERVER port- do not change switchport access vlan 103 switchport mode access duplex half spanning-tree portfast ! interface FastEthernet1/0/23 description HQ PHONE 2- 7962 phone switchport access vlan 101 switchport mode access switchport voice vlan 102 switchport voice detect cisco-phone spanning-tree portfast ! interface FastEthernet1/0/24 description *** DO NOT CHANGE - THIS IS YOUR L3 CONNECTION TO YOUR VPN!!! *** switchport access vlan 101 switchport mode access speed 100 duplex full no cdp enable ! interface GigabitEthernet1/0/1 ! interface GigabitEthernet1/0/2 ! --More-- interface Vlan1 no ip address ! interface Vlan101 ip address 10.10.100.3 255.255.255.0 ! interface Vlan102 ip address 10.10.200.120 255.255.255.0 ! control-plane ! ! line con
Re: [OSL | CCIE_Voice] B-ACD
I don't have a lab to test now, but this might be useful for your further investigation. From this link, http://www.cisco.com/en/US/docs/ios/voice/command/reference/vr_s01.html#wp1293780 It is interesting that the actual syntax to refer to built-in app is below: (I will test it to see if it works as soon as I get back to my lab) Router(config)# application Router(config-app)# service queue builtin:app-b-acd If it works, then I am pretty sure that we can do application no service queue builti:app-b-acd *What I don't know is somehow we are able to specify the service name without location and simply refer to the service name as the same name as the builtin app's name. From the look of it, the syntax we used according to the command reference is not even correct. I can only guess that it could be a bit of a change in IOS syntax over time. * * * *Another thing that may be worth trying is to remove all of the parameters from under the application itself to see if it will somehow remove that service from the startup/running configuration. * * * * * --Somphol On Sat, Jun 22, 2013 at 9:51 PM, CISCO CCIE VOICE ccievoic...@gmail.comwrote: Thanks Bill for your reply, I have done no service app-b-acd and no service app-b-acd-aa but showing all those commands in Running configuration thanks On Sat, Jun 22, 2013 at 1:48 PM, Bill Lake whl...@gmail.com wrote: If it is showing up in the running configuration, then you most likely see something like below, the best way to remove this is to no the commands Or to have done a Archive or copy of the config before you apply it. then restore that config as the startup and reboot. application * service app-b-acd * param number-of-hunt-grps 2 param aa-hunt2 param aa-hunt3 1222 param queue-len 15 param queue-manager-debugs 1 ! * service app-b-acd-aa * paramspace english index 1 paramspace english language en paramspace english location flash: param service-name app-b-acd param handoff-string app-b-acd-aa param aa-pilot 8005550123 param welcome-prompt _bacd_welcome.au param number-of-hunt-grps 2 param dial-by-extension-option 1 param second-greeting-time 60 param call-retry-timer 15 param max-time-call-retry 700 param max-time-vm-retry 2 param voice-mail 5003 ! dial-peer voice 222 voip service app-b-acd-aa destination-pattern 8005550123 session target ipv4:192.168.1.1 incoming called-number 8005550123 dtmf-relay h245-alphanumeric codec g711ulaw no vad On Sat, Jun 22, 2013 at 2:25 AM, Somphol Boonjing somp...@gmail.comwrote: That one is the embedded one so you actually can not remove it. However, you can simply ignore it and use one that is external script. So, if you have the external BACD script, you can use it instead of the embedded one. Branch2#show flash | inc bacd 107 30421bacd/app-b-acd-3.0.0.2.tcl 108 55599bacd/app-b-acd-aa-3.0.0.2.tcl application service *funnyqueue flash:/bacd/*app-b-acd-3.0.0.2.tcl -- you can you whatever name you like, in this case funnyqueue -- point the script to the script with correct path . (detail remove for brevity)... ! service* funnyaa flash:/bacd/app-b-acd-aa-3.0.0.2.tcl * -- you can you whatever name you like, in this case funnyaa -- point the script to the script with correct path . (detail remove for brevity). param service-name *funnyqueue* -- refer to your queue application name param handoff-string *funnyaa* . (detail remove for brevity). ! dial-peer voice 222 voip service *funnyaa* -- refer to your AA application name. . (detail remove for brevity)... ! To remove it from the running config, then you can, application no service *funnyqueue flash:/bacd/*app-b-acd-3.0.0.2.tcl no service* funnyaa flash:/bacd/app-b-acd-aa-3.0.0.2.tcl* Ref: http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html#wp1026305 Compared to using the embedded one below: application service app-b-acd -- you can't change the name of the embedded BACD Queue script . (detail remove for brevity)... ! service app-b-acd-aa -- you can't change the name of the embedded BACD AA script . (detail remove for brevity). param service-name app-b-acd -- refer to the embedded BACD Queue script param handoff-string app-b-acd-aa . (detail remove for brevity). ! dial-peer voice 222 voip service app-b-acd-aa -- refer to the name of the embedded BACD AA script . (detail remove for brevity)... ! Ref: Embedded Call-Queue and AA Tcl Scripts: Example http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html On 22/06/2013 4:18 PM, CISCO CCIE VOICE ccievoic...@gmail.com wrote
Re: [OSL | CCIE_Voice] UCM DHCP
Hi Anuritha, The screenshot is useful. The 4 parameters in the bottom of the screen must be set. Once you know what they means it will make sense. For now set them to in these order, you can change them later to fit your need. 3000 28800 14400 18800 On 23/06/2013 4:33 AM, anuritha konjety anurith...@gmail.com wrote: Whats interesting is if I do a sh cdp neigh detail on Switch A, I see one of the phones got an IP Device ID: SEP001BD4C6986C Entry address(es): IP address: 10.10.200.129 Platform: Cisco IP Phone 7960, Capabilities: Host Phone Interface: FastEthernet1/0/2, Port ID (outgoing port): Port 1 Holdtime : 138 sec But the phone itself doesn't seem to have an IP, nothing shows up under sh ip int bri either. Got a capture from callmanager pub. Filter to ip.addr == 10.10.200.3, I see DHCP request coming from 10.10.200.3 also see call manager responding with an IP(10.10.200.129). However the see the 10.10.200.3 send the request again again, looks like its stuck in a loop. Is this a bug? Attached to this email are screenshot f UCM config, capture from UCM pub(DHCP server) and sh run/debugs from Switch A, Router A. \itha konjety anurith...@gmail.com wrote: Debugs from Router: seeing similar errors //no SVI on switch// Jun 22 17:01:50.907: DHCPD: checking for expired leases. --More-- Jun 22 17:01:52.723: DHCPD: Sending notification of DISCOVER: Jun 22 17:01:52.723: DHCPD: htype 1 chaddr 001b.d4c6.986c Jun 22 17:01:52.723: DHCPD: remote id 020a0a0ac8030066 Jun 22 17:01:52.723: DHCPD: circuit id Jun 22 17:01:52.723: DHCPD: Seeing if there is an internally specified pool class: Jun 22 17:01:52.723: DHCPD: htype 1 chaddr 001b.d4c6.986c Jun 22 17:01:52.727: DHCPD: remote id 020a0a0ac8030066 Jun 22 17:01:52.727: DHCPD: circuit id Jun 22 17:01:52.727: DHCPD: setting giaddr to 10.10.200.3. Jun 22 17:01:52.727: DHCPD: BOOTREQUEST from 0100.1bd4.c698.6c forwarded to 10.10.210.10. On Sat, Jun 22, 2013 at 9:30 AM, anuritha konjety anurith...@gmail.comwrote: Thanks Somphol, I will start debugging at the router level. The SVI was initially not there, I was trying few different things to get this to work added it. On Sat, Jun 22, 2013 at 5:36 AM, Somphol Boonjing somp...@gmail.comwrote: A few points that I think worth double check: [1] Assuming this is the configuration of the DHCP on CUCM, I think the primary router doesn't require subnet mask to be specified DHCP Server : 10.10.210.10 subnet IPV4 address: 10.10.200.0 primary start addr: 10.10.200.120 primary end addr : 10.10.200.130 primary router : 10.10.200.3/255.255.255.0 [2] In the same broadcast domain (VLAN 102), you seem to have two routing interface, one on the router and another on as SVI on the switch.Unless it serves other purpose, I think you can safely remove the SVI. interface FastEthernet0/0.102 encapsulation dot1Q 102 ip address 10.10.200.3 255.255.255.0 ip helper-address 10.10.210.10 ! interface Vlan102 ip address 10.10.200.120 255.255.255.0 ! [3] Once the SVI interface vlan102 is removed, then the interface that actually forward your request to the CUCM DHCP should be the fa0/0.102 on the router that has already been configured with ip helper-address 10.10.210.10 (assuming that this is your CUCM IP). Then, the following trace on the switch will not be relevant anymore. 01:58:00: DHCPD: Finding a relay for client 0100.1bd4.c698.6c on interface Vlan102. 01:58:00: DHCPD: there is no pool for 10.10.200.120. Once removing SVI 102, then you should try to focus your debugging effort on the Router and not on the switch. [4] Your DHCP range seems to start with 10.10.200.120 and that could cause a collision with the SVI's IP address, even if everything else is correct. I remember that IOS-based DHCP seems to assign the last IP address in the pool, but I'm not sure how CUCM DHCP select the IP address from the pool. --Somphol On Sat, Jun 22, 2013 at 10:27 AM, anuritha konjety anurith...@gmail.com wrote: Ovidiu, Yes, ip helper-address is configured. Regards, Anu On Fri, Jun 21, 2013 at 4:32 PM, Ovidiu Popa ovi.p...@gmail.comwrote: Hello Anuritha Do you by any chance have an SVI on vlan 102 ? If yes have you tried configuring ip helper-address on that SVI ? Regards, Ovidiu On Sat, Jun 22, 2013 at 12:26 AM, anuritha konjety anurith...@gmail.com wrote: Hello, I am having trouble with getting phones Ip address when UCM(pub) is configured as the DHCP server. Following is partial config from the switch the router. I have made sure - 1. Dot1q trunk on HQ-RTR interface (on the switch). Make sure that all appropriate vlans are allowed on the trunk link and that native VLAN lines up (1 is default). 2. Ensure VLANs are provisioned correctly, assigned to the right interfaces, and active (sh vlan b) 3. Double check scope config on CUCM Pub. Check
Re: [OSL | CCIE_Voice] SRST checklist
To add to your list that is already good, - date/time format - timezone - system message - number of max calls busy call triggers - call pickup behavior if applicable (directed vs no directed call pickup) - call-transfer pattern - call-forward pattern - number of channels for ephone (dual, octal) - cbarge for shared line if applicable - SRST for media resources (via sdspfarm) - cptone if applicable --Somphol. --Somphol On Fri, Jun 21, 2013 at 3:08 PM, Karen Johnson karen.johnson...@yahoo.cawrote: all, i am trying to compile SRST check for my next attempt. I never got full mark here in my few attempts and always curious what i missing (even it seems I already done what they asked) - caller id and name ( hide or display) - Mwi light and VM message from PSTN and IP phone - inter site call VM inter site - COR if any - When forward call come , it play personal greeting - DND to divert - huntstop channel - if agents still working - after back to normal mode, verify everhthing -softkeys - feature : conference - timeout interdigit and Cfwd timer similar to UCM mode - always use preference 9 and dial-peer hunt 2 Any other tips and trick that I am not aware ? help please K - ___ For more information regarding industry leading 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] Correct order for Directory services
I've just tested this out. IP Service pointing to http://TFTPSERVER:6970/servicename.xml doesn't quite work for me, the phone reported XML Error [4]: Parse Error. I can retrieve the XML file using web browser. The same file hosted on UCCX's IIS has no problem. It might work with the newer CUCM/Phone firmware version, but I haven't tried that out yet. --Somphol. --Somphol On Mon, Jun 17, 2013 at 2:18 PM, Brian Meade bmead...@vt.edu wrote: Robert, HTTP port 6970 works fine on my 7.0 base cluster. It's been a hidden feature of the TFTP service for a while. A lot of the newer phones use this as you said for downloading anything TFTP used to be used for. One of the best features is the file list. No more switching to OS Admin and you don't have to deal with the case-sensitive search used by TFTP file management. Ctr+f is way easier/faster. http://x.x.x.x:6970/filelist.txt Brian Meade Date: Sun, 16 Jun 2013 20:46:02 -0700 From: Robert Thomas tho...@gmail.com To: Bill Lake whl...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Correct order for Directory services Message-ID: CAJ2RBBCCGkKhavv5KZ0W7heidG7= ycemui8sghsnal_uhpt...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 6970 is used with firmware 9.X and CUCM 8.5. Most of 69XX phones nowadays use HTTP instead of TFTP to download their files. Not sure if you would be able to get 6970 to work on version 7.X for the lab though. Nice to see Randall Again ;) On Sun, Jun 16, 2013 at 9:32 AM, Bill Lake whl...@gmail.com wrote: Did you test this and find it to work? The other day I remember seeing someone mentioned something about accessing TFTP files on CUCM using http URL. I think that is precious. So, the 'directory.xml' (or 'deny.xml') potentially can be hosted on the CUCM TFTP server itself. (I haven't tested that yet though) I think the URL is at the port 6970. (I memorize it as an adjacent number 69 and 70). http://CUCMFTP:6970/directory.xml Note: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/port/8_5_1/portlist851.html On Fri, Jun 7, 2013 at 6:50 PM, Somphol Boonjing somp...@gmail.com wrote: I love the detail step that Bill outlined above.+1 for that. The other day I remember seeing someone mentioned something about accessing TFTP files on CUCM using http URL. I think that is precious. So, the 'directory.xml' (or 'deny.xml') potentially can be hosted on the CUCM TFTP server itself. (I haven't tested that yet though) I think the URL is at the port 6970. (I memorize it as an adjacent number 69 and 70). http://CUCMFTP:6970/directory.xml Note: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/port/8_5_1/portlist851.html Centralized TFTP Alternate TFTP 6970 / TCP Centralized TFTP File Locator Service Regards, --Somphol On Sat, Jun 8, 2013 at 4:52 AM, Randall Saborio ill2...@gmail.com wrote: Hi Bill, Thanks a lot for the thorough steps. I can see very well how that should work and I will totally use that method if I am faced with the question. But then I take it the SQL insertion and use of priority field just doesn't work how I would expect. Any witnesses on using the SQL method and priority successfully?Just curious but may just as well ditch that method and use the custom XML as suggested by Bill. I always get the services added but always show up in alphabetical order. Cheers! On Fri, Jun 7, 2013 at 5:17 AM, Bill Lake whl...@gmail.com wrote: Ordering Directory Services Copy the Directory Services to be deleted or just uncheck enabled Device / Device Settings / Phone Services Enter each phone service and copy information for later use (should look like this Application:Cisco/CorporateDirectory) Uncheck or Delete the Directory Services Device / Device Settings / Phone Services Delete services or uncheck enable (uncheck is my preferred method) You should now have a list that looks like this (no I did not cut and paste each one so don't use this) Application:Cisco/CorporateDirectory Application:Cisco/MissedCalls Application:Cisco/PersonalDirectory Application:Cisco/PlacedCalls Now log into UCCX (or other web server but that might need some tweaking to make this work) and create an XML page, to do that go Cisco web page for support Products-Voice and Unified Communications-IP Telephony-Unified Communication Platform-Cisco Unified Communication Manager (CallManager) Programing Guides Cisco Unified IP Phone Services Application Development CiscoIPPhone XML Object Quick Reference Edit placing services in order desired save as directory.xml Paste it to c:\inetpub\wwwroot\directory.xml Change the Enterprise Parameters for URL Directories to http://IP UCCX/directory.xml Change Service Parameter for Directory
Re: [OSL | CCIE_Voice] UCCX Native Codec G729
Hi, I think the easiest way is to check UCCX Service Parameters under System menu. --Somphol. --Somphol On Sun, Jun 16, 2013 at 7:36 PM, CISCO CCIE VOICE ccievoic...@gmail.comwrote: Hi, How can i verify that UCCX is using G729 codec native Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Lab strategy
Just my personal experience when I failed my first lab attempt. CUE is at the forefront of the time waster. Simply because it requires constant reboot if mistake is made and every reboots take a long time too. 10 ) CUE integration and trafer setup takes 20 mins My brain seems to go numb at times too. Simple stuff becomes a lot harder, and I still looking back on the day as how on earth I miss that easy fix. Not saying that without it I would pass, I just think I wouldn't the thoroughly outclassed by it. Regards, --Somphol. --Somphol On Thu, Jun 6, 2013 at 2:17 AM, Martin Sloan martinsloa...@gmail.comwrote: Let me say, I've never sat the lab so I'm just commenting on my own study experience. - Points 3 4 are probably consuming too much time. Documenting IP's is good but I don't currently go through and check configs. I figure i'll get to it when it's time to configure that piece and if there are issues I'll do my troubleshooting then. - 1.5hrs for Br2 seems like a long time. I think you can get quicker with that part and do it in 45min or less, depending on the complexity. Just with those savings you're getting closer to your 6 hour goal. I use notepad for anything that can be repeated like dial-peers, translation rules/patterns, dspfarm, QoS, etc. I think its MUCH faster than manually typing the configs each time. I always configure BR1 first so most of the time I'm just tweaking some settings and pasting into BR2. Just my input based on my own studies. I'm sure there are veterans out there who have a better insight into saving time. I'm interested to hear their take as well. Marty On Wed, Jun 5, 2013 at 10:58 AM, singh singh8...@in.com wrote: hi Everyone, I need your inputs here. I have been trying to complete mock and practice labs in 8 hours . However their I am unable to finish or I finish with lab with a lot of mistakes with no time for testing. I also realize that I lose thing in the first half and speed up during the end . Generally I take... 1) 10 mins to test if all equipment is working fine ( 10 mins) 2) read the workbook questions for the next 15 mins 3) Make note of the ip addresses and router configuration per site in 25 mins 4) Make note of all cucm configuration , cups , unity connection , uccx and cue for another dial plan 20 mins 5) Now from point 5 - I start with lab configurations from Branch 2 ( site c - which is a mgcp gateway and srst setup) this generally takes me about 1 and a half hour to just complete all configuration ( 1 hour and 30 mins) 6) Then I move to Branch 1 ( site B - which is a H323 gateway with srst) this ge nerally takes about 45 - 50 mins 7) I then move to HQ ( R1 - which is MGCP gateway with srst ) this generally takes 20 mins . 8) Basic setup of DP , css, aar, NTP , service parameters and enterprise para , vlans , dhcp and ip phone registration takes 50 mins 9) CUC integ and other config including recording takes 20 mins 10 ) CUE integration and trafer setup takes 20 mins 11) UCCX integration , One button , script and recording takes 40 mins 12) CUPs integration and client setup takes 20 mins 13 )I then come back to callmanger and do the media resource setup , gateways added cucm , other configuration such as MVA , RSVP , + dial , adding trunks , unassigned dn setup to CUC - this takes 50 mins 14 ) Then I move to the Route pattern setup on callmanger this I do for 3 sites - HQ , Site B and Site C with or without redundancy on callmanger this takes 25 mins 15 ) Then do this such as RTMT log collection and indicating informa tion in seperate files , MGCP debugs this takes another 15 mins 16 ) Switch and WAN QOS this I plan to do only if time permits as this is a complex section Questions : === 1) I barely am able to finish things in time . I have heard from on this forum that there are candidates who finish it is 5 - 6 hours . Would anyone be able to share with me as to how they do this ? 2) Even if the above I complete exercises are complete there are sections where I miss out on configurations. How do I make sure all config for all sections is done correctly? 3) I really wish I am able to finish the lab in 6 hours so that I can test for another 2 hours . Could someone therefore check the above 16 points and let me know about the time I can reduce. 4) As you can see above the router configurations consume a bit more time making ( points 5 , 6 , 7 ) . I have tried with both using a notepad to type the configurations and then paste also with typing on cli but both these methods take around the same time. Please let me know what best method I can use for points 5 , 6 , 7. 5) Other suggestions are most welcome . Thanks guys in advance for all your help! Regards, Singh Get Yourself a cool, short *@in.com* Email ID now!http://www3.in.com/sso/commonregister.php?ref=INutm_source=inviteutm_medium=outgoing
Re: [OSL | CCIE_Voice] Lab strategy
Thanks, Bill for great information. BTW, if I could ask about your thought on the VNC-only workstation. I don't really understand the logic behind making and RDP available for Windows-based UCCX server, and only supply the VNC session for another utility host. (I was in a rush to such an extent that I couldn't even have time nor effort to spend on finding the password to access the VNC session, I know it is right there somewhere obvious. Sigh. Hope I do better next time.) Regards, --Somphol. --Somphol On Thu, Jun 6, 2013 at 10:04 AM, Bill Lake whl...@gmail.com wrote: Where are you taking your lab? That has to play into your plan in taking the lab. I took my lab at RTP, my plan was to use the device base approach. 1) I reviewed the outline of the lab, looking for very specific things throughout the lab that required early planning. An example of this would be a System wide parameter that I would need to configure early on. 2) quickly notepad the items I would need throughout the lab 3) rough scope of the lab, my device based approach and dial plan page This took about 20 minutes to complete and in my opinion you should not spend more than 30 and your goal should be 15. At this point I started my device based approach and for me this was my order 1) HQ switch 2) do base to get CUE started and start CUE 3) HQ Router 4) SB router 5) Check on CUE, do anything I thought I needed and could then reboot it 6) SC router 7) finish CUE 8) base CUCM 9) dial-plan with testing as I go all calls in round robin so I am using every phone to make calls (try to complete to this point) when 5 minutes before archive or back up all configs you can just incase Lunch 10) finish CUCM 11) CUC 12) CUPS 13) UCCX 14) Test everything you can Make sure you do not get stuck on one thing, spend 10 minutes and move on, circle back and try to complete it but keep moving. I know it sounds simple but you will be amazed how long you might spend on a 2-3 point item you know you can do when it is costing you other points you could be scoring. During your testing focus on being the proctor and thinking how he might find fault in what you did. Almost working perfectly won't get you the points. Bill On Wed, Jun 5, 2013 at 11:17 AM, Martin Sloan martinsloa...@gmail.comwrote: Let me say, I've never sat the lab so I'm just commenting on my own study experience. - Points 3 4 are probably consuming too much time. Documenting IP's is good but I don't currently go through and check configs. I figure i'll get to it when it's time to configure that piece and if there are issues I'll do my troubleshooting then. - 1.5hrs for Br2 seems like a long time. I think you can get quicker with that part and do it in 45min or less, depending on the complexity. Just with those savings you're getting closer to your 6 hour goal. I use notepad for anything that can be repeated like dial-peers, translation rules/patterns, dspfarm, QoS, etc. I think its MUCH faster than manually typing the configs each time. I always configure BR1 first so most of the time I'm just tweaking some settings and pasting into BR2. Just my input based on my own studies. I'm sure there are veterans out there who have a better insight into saving time. I'm interested to hear their take as well. Marty On Wed, Jun 5, 2013 at 10:58 AM, singh singh8...@in.com wrote: hi Everyone, I need your inputs here. I have been trying to complete mock and practice labs in 8 hours . However their I am unable to finish or I finish with lab with a lot of mistakes with no time for testing. I also realize that I lose thing in the first half and speed up during the end . Generally I take... 1) 10 mins to test if all equipment is working fine ( 10 mins) 2) read the workbook questions for the next 15 mins 3) Make note of the ip addresses and router configuration per site in 25 mins 4) Make note of all cucm configuration , cups , unity connection , uccx and cue for another dial plan 20 mins 5) Now from point 5 - I start with lab configurations from Branch 2 ( site c - which is a mgcp gateway and srst setup) this generally takes me about 1 and a half hour to just complete all configuration ( 1 hour and 30 mins) 6) Then I move to Branch 1 ( site B - which is a H323 gateway with srst) this ge nerally takes about 45 - 50 mins 7) I then move to HQ ( R1 - which is MGCP gateway with srst ) this generally takes 20 mins . 8) Basic setup of DP , css, aar, NTP , service parameters and enterprise para , vlans , dhcp and ip phone registration takes 50 mins 9) CUC integ and other config including recording takes 20 mins 10 ) CUE integration and trafer setup takes 20 mins 11) UCCX integration , One button , script and recording takes 40 mins 12) CUPs integration and client setup takes 20 mins 13 )I then come back to callmanger and do the media resource
