Re: [OSL | CCIE_Voice] WB1 LAB4A - Task 4.6 Gatekeeper
hi, I always thought that this number results from the order of CCMs in the CUCM group assigned to the trunk. But if you have only 2 CCMs in the group (have you checked this? maybe the previous subscriber installation was left there and is inactive somehow?), then this number is indeed taken from the order in which a server were added to the cluster. regards On Sun, Jul 4, 2010 at 4:45 PM, Duncan Hamilton-Walker dun...@rosethorn.plus.com wrote: Hi Matt, Yes i’m using my own lab.. thinking about it..the SUB has been rebuilt, due to an issue with the DB.. So im thinking that the PUB thinks that this is subscriber 2.. when its actually subscriber 1 rebuilt.. hence giving it an identifier of _3 Would that make sense.. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SRST
hi, doesn't this break the 2nd requirement? I've never tried this, but I would configure ephone-template and assign it to srst via srst ephone template command of telephony-services. sccp group should indeed be configured with srst as the 3rd option. regards kobel On Fri, Jul 2, 2010 at 4:29 PM, Ashar Siddiqui siddas...@gmail.com wrote: Sean, Do srst auto-prov none and then just create ephones (as many as required) and put the following in there: Ephone 1 No privacy ! Ephone 2 No privacy ! You will need to do all the basic requirements for Cbarge like conference hardware, sdspfarm units etc and configuring dspfarm with telephony-service address on third priority if the requirement is that Cbarge is working during normal mode as well. Give it a go and let us know how it works. Ash ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Presence Issues Softphone mode in CUPC Lab 5 Volume 2
That's right, the CUPC device name is constructed from first 12 characters of the username - so the usernames longer then 12 characters are acceptable, but are trimmed (the get the same lenght as IP phone name SEP + 12 characters of MAC address). thanks for the info about user groups needed to make it work - I always did it. However, Owner User ID is pointed in the official configuration guide. I love CUPS documentation. regards On Sun, Jun 27, 2010 at 3:45 AM, Daniel Berlinski dberlin...@gmail.comwrote: Hi Guys I want to let you all know that it works only if the username is lower then 12 characters and match exactly with the UPCdevicename. Also for you info, phone Owner ID and assigning the end user to the CCM End users and CTI enabled groups was not neceessary to make it work. Cheers On Sun, Jun 27, 2010 at 12:16 PM, Daniel Berlinski dberlin...@gmail.comwrote: Hello Pavan CUPC is not even requesting the config xml file, checked with wireshark. In show server health there is no value against TFTP.Filename= I can't get it to work even after the client upgrade. I guess I will re-image the CUPS server and will update later. Cheers 2010/6/27 Pavan pav.c...@gmail.com Daniel, Before you go check replication, check to see if cups is even requesting the correct config xml file. Replication could only be a problem when cups tries to register to ucm and ucm rejects the register request Sent from my phone On Jun 26, 2010, at 16:52, Roger Källberg roger.kallb...@cygate.se wrote: Hi Daniel, It's not always that you can trust the information given by the show perf query class Number of Replicates Created and State of Replication command. One easy thing that you can do to verify if you have a db repl problem is to put your phones, or any other device, in a pub only enviroment. If all works then you know that the sub didn't have the correct info. And in thet case you need to repair the db replication by utils debreplication stop ,1 on sub, then when promtpt returns on the sub put in the same command on pub). When the prompt returns on the pub use utils dbreplication repair all on the pub. This will take some time to complete. *Roger Källberg* CCIE #26199 (Voice) Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ Direkt: +46108787498 Växel: +46108787400 roger.kallb...@cygate.se -- *Från:* Daniel Berlinski [dberlin...@gmail.com] *Skickat:* den 26 juni 2010 23:44 *Till:* Roger Källberg *Kopia:* kobel; osl osl *Ämne:* Re: [OSL | CCIE_Voice] Presence Issues Softphone mode in CUPC Lab 5 Volume 2 Hi Kobel Owner was setup for the mobility section to work. It is in there. Hi Roger The way I know how to verify dbReplication is: admin:show perf query class Number of Replicates Created and State of Replication ==query class : - Perf class (Number of Replicates Created and State of Replication) has instances and values: ReplicateCount - Number of Replicates Created = 412 ReplicateCount - Replicate_State= 2 My reading of this is that is all good. Am I right? Well, I have rebooted this many times already so I think I will just upgrade the client and see what happens. Will update you all. Thnaks 2010/6/27 Roger Källberg roger.kallb...@cygate.se roger.kallb...@cygate.se Try to verify if db replication is ok, if not, fix that. You might also want to restart the CTI Manager on both sub and pub. Brgds, *Roger Källberg* CCIE #26199 (Voice) Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ Direkt: +46108787498 Växel: +46108787400 roger.kallb...@cygate.se -- *Från:* Daniel Berlinski [ dberlin...@gmail.comdberlin...@gmail.com] *Skickat:* den 26 juni 2010 23:18 *Till:* kobel *Kopia:* osl osl *Ämne:* Re: [OSL | CCIE_Voice] Presence Issues Softphone mode in CUPC Lab 5 Volume 2 Thanks for your replies. Primary extension is assigned to end user and that extension matches with the line number of CUPC. The users are assigned to the Standard CCM End Users, and CTI Enabled groups What is the version of CUPC you guys use? Thank you On Sun, Jun 27, 2010 at 9:03 AM, kobel findko...@gmail.com findko...@gmail.com wrote: See if adding the end user to Standard CUCM users group in CUCM helps regards On Sat, Jun 26, 2010 at 10:30 PM, Daniel Berlinski dberlin...@gmail.com dberlin...@gmail.com wrote: Hello all Out of ideas now after troubleshooting extensively a Presence problem. I'm finalizing lab 5 Vol 2 and I can't get CUPC to download its TFTP configuration file from CUCM and for that reason I do not even see the option for selecting softphone control Any help is appreciated. What I have and what I've done is the following: 1- Cretaed device named UPC+12alphanumeric characters, in my case UPCTERRELLEPRYO, associated its line to the enduser 2- End user configured with primary
Re: [OSL | CCIE_Voice] Presence Issues Softphone mode in CUPC Lab 5 Volume 2
I was trying to remind myself why I always did it this way. It might be needed if you want CUPC to register to CUPS without adding ALL ACL in CUPS (this is how proctor guide handles this). The alternative way is to configure CUPS device with Owner User ID and Digest User ID + digest credentials for this user. The CUPC downloads configuration file with digest credentials and uses them when registering with CUPC. It's not documented very well. On Sun, Jun 27, 2010 at 11:42 AM, kobel findko...@gmail.com wrote: That's right, the CUPC device name is constructed from first 12 characters of the username - so the usernames longer then 12 characters are acceptable, but are trimmed (the get the same lenght as IP phone name SEP + 12 characters of MAC address). thanks for the info about user groups needed to make it work - I always did it. However, Owner User ID is pointed in the official configuration guide. I love CUPS documentation. regards On Sun, Jun 27, 2010 at 3:45 AM, Daniel Berlinski dberlin...@gmail.comwrote: Hi Guys I want to let you all know that it works only if the username is lower then 12 characters and match exactly with the UPCdevicename. Also for you info, phone Owner ID and assigning the end user to the CCM End users and CTI enabled groups was not neceessary to make it work. Cheers On Sun, Jun 27, 2010 at 12:16 PM, Daniel Berlinski dberlin...@gmail.comwrote: Hello Pavan CUPC is not even requesting the config xml file, checked with wireshark. In show server health there is no value against TFTP.Filename= I can't get it to work even after the client upgrade. I guess I will re-image the CUPS server and will update later. Cheers 2010/6/27 Pavan pav.c...@gmail.com Daniel, Before you go check replication, check to see if cups is even requesting the correct config xml file. Replication could only be a problem when cups tries to register to ucm and ucm rejects the register request Sent from my phone On Jun 26, 2010, at 16:52, Roger Källberg roger.kallb...@cygate.se wrote: Hi Daniel, It's not always that you can trust the information given by the show perf query class Number of Replicates Created and State of Replication command. One easy thing that you can do to verify if you have a db repl problem is to put your phones, or any other device, in a pub only enviroment. If all works then you know that the sub didn't have the correct info. And in thet case you need to repair the db replication by utils debreplication stop ,1 on sub, then when promtpt returns on the sub put in the same command on pub). When the prompt returns on the pub use utils dbreplication repair all on the pub. This will take some time to complete. *Roger Källberg* CCIE #26199 (Voice) Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ Direkt: +46108787498 Växel: +46108787400 roger.kallb...@cygate.se -- *Från:* Daniel Berlinski [dberlin...@gmail.com] *Skickat:* den 26 juni 2010 23:44 *Till:* Roger Källberg *Kopia:* kobel; osl osl *Ämne:* Re: [OSL | CCIE_Voice] Presence Issues Softphone mode in CUPC Lab 5 Volume 2 Hi Kobel Owner was setup for the mobility section to work. It is in there. Hi Roger The way I know how to verify dbReplication is: admin:show perf query class Number of Replicates Created and State of Replication ==query class : - Perf class (Number of Replicates Created and State of Replication) has instances and values: ReplicateCount - Number of Replicates Created = 412 ReplicateCount - Replicate_State= 2 My reading of this is that is all good. Am I right? Well, I have rebooted this many times already so I think I will just upgrade the client and see what happens. Will update you all. Thnaks 2010/6/27 Roger Källberg roger.kallb...@cygate.se roger.kallb...@cygate.se Try to verify if db replication is ok, if not, fix that. You might also want to restart the CTI Manager on both sub and pub. Brgds, *Roger Källberg* CCIE #26199 (Voice) Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ Direkt: +46108787498 Växel: +46108787400 roger.kallb...@cygate.se -- *Från:* Daniel Berlinski [ dberlin...@gmail.comdberlin...@gmail.com] *Skickat:* den 26 juni 2010 23:18 *Till:* kobel *Kopia:* osl osl *Ämne:* Re: [OSL | CCIE_Voice] Presence Issues Softphone mode in CUPC Lab 5 Volume 2 Thanks for your replies. Primary extension is assigned to end user and that extension matches with the line number of CUPC. The users are assigned to the Standard CCM End Users, and CTI Enabled groups What is the version of CUPC you guys use? Thank you On Sun, Jun 27, 2010 at 9:03 AM, kobel findko...@gmail.com findko...@gmail.com wrote: See if adding the end user to Standard CUCM users group in CUCM helps regards On Sat, Jun 26, 2010 at 10:30 PM, Daniel Berlinski dberlin...@gmail.com
Re: [OSL | CCIE_Voice] Presence Issues Softphone mode in CUPC Lab 5 Volume 2
I did it some time ago and I think it worked for me (I might have used different username, but probably used the same - I can't verify this right now). Did you put a space after UPC on purpose? Normally you shouln't have any space in the device name, i.e.: UPCTERRELLEPRYO On Sun, Jun 27, 2010 at 5:29 PM, Daniel Berlinski dberlin...@gmail.comwrote: Hi there again Not sure if you did lab 5 of volume 2 but when you mentioned in your e-mail (the get the same lenght as IP phone name SEP + 12 characters of MAC address) never worked for me. I had the following users in CUCM: Terrell Pryor and Jake Stoneburner - These users with their UPC + 12 characters of their username:UPC TERRELLEPRYO for example never worked. Did you have the same experience while doing that lab? Thanks and apologies for being repetitive but it took me so long to do this lab that I want to ensure I did not miss anything. Cheers On Sun, Jun 27, 2010 at 9:42 PM, kobel findko...@gmail.com wrote: That's right, the CUPC device name is constructed from first 12 characters of the username - so the usernames longer then 12 characters are acceptable, but are trimmed (the get the same lenght as IP phone name SEP + 12 characters of MAC address). thanks for the info about user groups needed to make it work - I always did it. However, Owner User ID is pointed in the official configuration guide. I love CUPS documentation. regards On Sun, Jun 27, 2010 at 3:45 AM, Daniel Berlinski dberlin...@gmail.comwrote: Hi Guys I want to let you all know that it works only if the username is lower then 12 characters and match exactly with the UPCdevicename. Also for you info, phone Owner ID and assigning the end user to the CCM End users and CTI enabled groups was not neceessary to make it work. Cheers On Sun, Jun 27, 2010 at 12:16 PM, Daniel Berlinski dberlin...@gmail.com wrote: Hello Pavan CUPC is not even requesting the config xml file, checked with wireshark. In show server health there is no value against TFTP.Filename= I can't get it to work even after the client upgrade. I guess I will re-image the CUPS server and will update later. Cheers 2010/6/27 Pavan pav.c...@gmail.com Daniel, Before you go check replication, check to see if cups is even requesting the correct config xml file. Replication could only be a problem when cups tries to register to ucm and ucm rejects the register request Sent from my phone On Jun 26, 2010, at 16:52, Roger Källberg roger.kallb...@cygate.se wrote: Hi Daniel, It's not always that you can trust the information given by the show perf query class Number of Replicates Created and State of Replication command. One easy thing that you can do to verify if you have a db repl problem is to put your phones, or any other device, in a pub only enviroment. If all works then you know that the sub didn't have the correct info. And in thet case you need to repair the db replication by utils debreplication stop ,1 on sub, then when promtpt returns on the sub put in the same command on pub). When the prompt returns on the pub use utils dbreplication repair all on the pub. This will take some time to complete. *Roger Källberg* CCIE #26199 (Voice) Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ Direkt: +46108787498 Växel: +46108787400 roger.kallb...@cygate.se -- *Från:* Daniel Berlinski [dberlin...@gmail.com] *Skickat:* den 26 juni 2010 23:44 *Till:* Roger Källberg *Kopia:* kobel; osl osl *Ämne:* Re: [OSL | CCIE_Voice] Presence Issues Softphone mode in CUPC Lab 5 Volume 2 Hi Kobel Owner was setup for the mobility section to work. It is in there. Hi Roger The way I know how to verify dbReplication is: admin:show perf query class Number of Replicates Created and State of Replication ==query class : - Perf class (Number of Replicates Created and State of Replication) has instances and values: ReplicateCount - Number of Replicates Created = 412 ReplicateCount - Replicate_State= 2 My reading of this is that is all good. Am I right? Well, I have rebooted this many times already so I think I will just upgrade the client and see what happens. Will update you all. Thnaks 2010/6/27 Roger Källberg roger.kallb...@cygate.se roger.kallb...@cygate.se Try to verify if db replication is ok, if not, fix that. You might also want to restart the CTI Manager on both sub and pub. Brgds, *Roger Källberg* CCIE #26199 (Voice) Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ Direkt: +46108787498 Växel: +46108787400 roger.kallb...@cygate.se -- *Från:* Daniel Berlinski [ dberlin...@gmail.comdberlin...@gmail.com ] *Skickat:* den 26 juni 2010 23:18 *Till:* kobel *Kopia:* osl osl *Ämne:* Re: [OSL | CCIE_Voice] Presence Issues Softphone mode in CUPC Lab 5 Volume 2 Thanks for your replies. Primary
Re: [OSL | CCIE_Voice] Lab 5 Volume 2 LLQ sizing and RSVP CAC
Hi, When calculating bandwidth needed for PQ, I wouldn't take into account the value used for initial call by RSVP. It's never actually used, it's only a worst case scenario. voice packets are small and are never fragmented by FRF.12. that's why additional 4B in header are not needed. On Sun, Jun 27, 2010 at 6:50 PM, Daniel Berlinski dberlin...@gmail.comwrote: Hello list Volume 2 lab 5 has a scenario asking us to allow for 4 concurrent g729 calls over Frame FRF.12 LFI using RSVP for CAC. Proctor Guide has calculated the size of the priority queue without taking into account that first call prior to capabilities exchange that RSVP negotiates at 40Kbps. In addition Proctor Guide has used Frame Relay payload of 4 Bytes instead of 8 Bytes for FR with LFI. I answered this question as follows: For 4 g729r8 concurrent calls over the WAN using RSVP for CAC: compressed ip/udp/rtp=2bytes FRF.12=8Bytes g729 payload @ 20ms=20Bytes 30*50*8/1000=12Kbps per call so 3 calls=36Kbps 1 call all @ worse case scenario compressed ip/udp/rtp=2bytes FRF.12=8Bytes g729 payload @ 10ms=10bytes 20*100*8/1000 = 1 call 16Kbps So 4 calls=36kbps + 16Kbps= 52Kbps configured in priority queue Can anyone let me know if my approach is right or wrong and if wrong why? Thanks a lot Daniel ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol2 Lab1 6.1 - Weird MOH from flash
dstCallID 103 Jun 26 00:51:14.655: moh_process_ccb: dstadr 10.30.200.68, callid 103, port 32410, codec 16, moh_en 0, moh_addr 0.0.0.0 Jun 26 00:51:14.655: moh_update_rtp: callID 102 dstCallID 103 Jun 26 00:51:15.607: moh_update_rtp: callID 102 dstCallID 103 Jun 26 00:51:15.615: moh_update_rtp: callID 102 dstCallID 103 Jun 26 00:51:15.647: %ISDN-6-CONNECT: Interface Serial0/2/0:2 is now connected to 3942123 N/A Jun 26 00:51:15.647: %ISDN-6-DISCONNECT: Interface Serial0/2/0:2 disconnected from 3942123 , call lasted 4 seconds HQ-RTR# Jun 26 00:51:15.651: ISDN Se0/2/0:23 Q931: TX - DISCONNECT pd = 8 callref = 0x00A8 Cause i = 0x8090 - Normal call clearing Jun 26 00:51:15.659: ISDN Se0/2/0:23 Q931: RX - RELEASE pd = 8 callref = 0x80A8 Jun 26 00:51:15.663: ISDN Se0/2/0:23 Q931: TX - RELEASE_COMP pd = 8 callref = 0x00A8 On Fri, Jun 25, 2010 at 7:27 PM, kobel findko...@gmail.com wrote: I see no other reason for MoH not working for specific PSTN lines. Make sure you understand your call routing correctly - e.g. debug isdn q931 to confirm that the calls go through the right gateway. Calls may be routed via different gateway that you think. Else: * Have you tried the service parameter I mentioned (on both MOH servers)? * Make sure your H.323 gateway is configured with voice class codec allowing g711 On Sat, Jun 26, 2010 at 2:00 AM, Tam Nhu tamnhu...@gmail.com wrote: Hi Kobel, Thanks for your input. All those codec, region, dp, etc are settings correctly, including the gateway. As I mentioned in Bold, *only call to that particular PSTN line #2 (HQ local PSTN # 212-394-2123) is having 'beep' on-hold. All other calls are working fine*...that's weird. I run the debug ccm-man mus all and see that for the bad 'beep' moh, it did not find the moh destination ip address. Below are the output samples of 'bad moh' (call to 3942123) and 'good moh' (call to 911) ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Presence Issues Softphone mode in CUPC Lab 5 Volume 2
See if adding the end user to Standard CUCM users group in CUCM helps regards On Sat, Jun 26, 2010 at 10:30 PM, Daniel Berlinski dberlin...@gmail.comwrote: Hello all Out of ideas now after troubleshooting extensively a Presence problem. I'm finalizing lab 5 Vol 2 and I can't get CUPC to download its TFTP configuration file from CUCM and for that reason I do not even see the option for selecting softphone control Any help is appreciated. What I have and what I've done is the following: 1- Cretaed device named UPC+12alphanumeric characters, in my case UPCTERRELLEPRYO, associated its line to the enduser 2- End user configured with primary extension, associated with UPC phone device, CTI control of its devices and group association to CTI enabled group. 3- Still in CUCM, Capabilities Assignment was provided for the user. 5- In CUPS, Application-Cisco Unified Personal Comm - Settings I have provided the IP addresses for TFTP server primary and secondary Presence status is working fine and Deskphone control works fine as well. My problem here is that the CUPC SIP phone is not getting in Show Server Health a tftp file to download. It displays the IP addres of TFTP primary and seoondary but it does not display the UPCTERRELLEPRYO.CNF.XML file to download. To troubleshoot this I have done the following: 1- Went in DOS and did a tftp -i 10.10.210.10 get UPCTERRELLEPRYO.CNF.XML and files downloaded OK so there is no network issues here. Inside the file I saw references to TFTP server as IP addresses so no name resolution issues either. 2- Ran Wireshark and did not see any attempts from the client machine to register with CUCM via SIP so client is not even attempting to register. In fact nothing displays when I filter the capture by the CUCM ip addresses. 3- Listing my cupc users by clicking in CUPS, application, Cisco Unified personal comm, user settings I see my users listed there but under the column Client Type nothing displays 4- Created another UPC device for another user with another name and it still presents same problem. 5- Tried to enable all phone tracing in CUCM and everything else related to SIP under trace settings and nothing displayed with relation to the UPC phone attempting to register. Not sure what to do next, my version of CUPC is 7.0(2.13496) I havem't looked for bugs yet. What version are you guys using? If anyone has any ideas please let me know ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Presence Issues Softphone mode in CUPC Lab 5 Volume 2
7.1(1.1237...). but there is one more thing - on the CUPC device configuration page - have you selected the correct owner user ID? On Sat, Jun 26, 2010 at 11:18 PM, Daniel Berlinski dberlin...@gmail.comwrote: Thanks for your replies. Primary extension is assigned to end user and that extension matches with the line number of CUPC. The users are assigned to the Standard CCM End Users, and CTI Enabled groups What is the version of CUPC you guys use? Thank you On Sun, Jun 27, 2010 at 9:03 AM, kobel findko...@gmail.com wrote: See if adding the end user to Standard CUCM users group in CUCM helps regards On Sat, Jun 26, 2010 at 10:30 PM, Daniel Berlinski dberlin...@gmail.comwrote: Hello all Out of ideas now after troubleshooting extensively a Presence problem. I'm finalizing lab 5 Vol 2 and I can't get CUPC to download its TFTP configuration file from CUCM and for that reason I do not even see the option for selecting softphone control Any help is appreciated. What I have and what I've done is the following: 1- Cretaed device named UPC+12alphanumeric characters, in my case UPCTERRELLEPRYO, associated its line to the enduser 2- End user configured with primary extension, associated with UPC phone device, CTI control of its devices and group association to CTI enabled group. 3- Still in CUCM, Capabilities Assignment was provided for the user. 5- In CUPS, Application-Cisco Unified Personal Comm - Settings I have provided the IP addresses for TFTP server primary and secondary Presence status is working fine and Deskphone control works fine as well. My problem here is that the CUPC SIP phone is not getting in Show Server Health a tftp file to download. It displays the IP addres of TFTP primary and seoondary but it does not display the UPCTERRELLEPRYO.CNF.XML file to download. To troubleshoot this I have done the following: 1- Went in DOS and did a tftp -i 10.10.210.10 get UPCTERRELLEPRYO.CNF.XML and files downloaded OK so there is no network issues here. Inside the file I saw references to TFTP server as IP addresses so no name resolution issues either. 2- Ran Wireshark and did not see any attempts from the client machine to register with CUCM via SIP so client is not even attempting to register. In fact nothing displays when I filter the capture by the CUCM ip addresses. 3- Listing my cupc users by clicking in CUPS, application, Cisco Unified personal comm, user settings I see my users listed there but under the column Client Type nothing displays 4- Created another UPC device for another user with another name and it still presents same problem. 5- Tried to enable all phone tracing in CUCM and everything else related to SIP under trace settings and nothing displayed with relation to the UPC phone attempting to register. Not sure what to do next, my version of CUPC is 7.0(2.13496) I havem't looked for bugs yet. What version are you guys using? If anyone has any ideas please let me know ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Presence Issues Softphone mode in CUPC Lab 5 Volume 2
yes, 2 means it's ok... you can try to restart the sync service on CUPS (in CUPS serviceability). and make sure the DN/partition pair is the same for deskphone and softphone. generally, review the steps in: http://docwiki.cisco.com/wiki/Cisco_Unified_Presence%2C_Release_7.x_--_How_to_Configure_Cisco_Unified_Personal_Communicator_on_Cisco_Unified_Communications_Manager#Creating_a_Softphone_Device_for_Each_Cisco_Unified_Personal_Communicator_User no more ideas - good luck and good night ;) 2010/6/26 Daniel Berlinski dberlin...@gmail.com Hi Kobel Owner was setup for the mobility section to work. It is in there. Hi Roger The way I know how to verify dbReplication is: admin:show perf query class Number of Replicates Created and State of Replication ==query class : - Perf class (Number of Replicates Created and State of Replication) has instances and values: ReplicateCount - Number of Replicates Created = 412 ReplicateCount - Replicate_State= 2 My reading of this is that is all good. Am I right? Well, I have rebooted this many times already so I think I will just upgrade the client and see what happens. Will update you all. Thnaks 2010/6/27 Roger Källberg roger.kallb...@cygate.se Try to verify if db replication is ok, if not, fix that. You might also want to restart the CTI Manager on both sub and pub. Brgds, *Roger Källberg* CCIE #26199 (Voice) Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ Direkt: +46108787498 Växel: +46108787400 roger.kallb...@cygate.se -- *Från:* Daniel Berlinski [dberlin...@gmail.com] *Skickat:* den 26 juni 2010 23:18 *Till:* kobel *Kopia:* osl osl *Ämne:* Re: [OSL | CCIE_Voice] Presence Issues Softphone mode in CUPC Lab 5 Volume 2 Thanks for your replies. Primary extension is assigned to end user and that extension matches with the line number of CUPC. The users are assigned to the Standard CCM End Users, and CTI Enabled groups What is the version of CUPC you guys use? Thank you On Sun, Jun 27, 2010 at 9:03 AM, kobel findko...@gmail.com wrote: See if adding the end user to Standard CUCM users group in CUCM helps regards On Sat, Jun 26, 2010 at 10:30 PM, Daniel Berlinski dberlin...@gmail.com wrote: Hello all Out of ideas now after troubleshooting extensively a Presence problem. I'm finalizing lab 5 Vol 2 and I can't get CUPC to download its TFTP configuration file from CUCM and for that reason I do not even see the option for selecting softphone control Any help is appreciated. What I have and what I've done is the following: 1- Cretaed device named UPC+12alphanumeric characters, in my case UPCTERRELLEPRYO, associated its line to the enduser 2- End user configured with primary extension, associated with UPC phone device, CTI control of its devices and group association to CTI enabled group. 3- Still in CUCM, Capabilities Assignment was provided for the user. 5- In CUPS, Application-Cisco Unified Personal Comm - Settings I have provided the IP addresses for TFTP server primary and secondary Presence status is working fine and Deskphone control works fine as well. My problem here is that the CUPC SIP phone is not getting in Show Server Health a tftp file to download. It displays the IP addres of TFTP primary and seoondary but it does not display the UPCTERRELLEPRYO.CNF.XML file to download. To troubleshoot this I have done the following: 1- Went in DOS and did a tftp -i 10.10.210.10 get UPCTERRELLEPRYO.CNF.XML and files downloaded OK so there is no network issues here. Inside the file I saw references to TFTP server as IP addresses so no name resolution issues either. 2- Ran Wireshark and did not see any attempts from the client machine to register with CUCM via SIP so client is not even attempting to register. In fact nothing displays when I filter the capture by the CUCM ip addresses. 3- Listing my cupc users by clicking in CUPS, application, Cisco Unified personal comm, user settings I see my users listed there but under the column Client Type nothing displays 4- Created another UPC device for another user with another name and it still presents same problem. 5- Tried to enable all phone tracing in CUCM and everything else related to SIP under trace settings and nothing displayed with relation to the UPC phone attempting to register. Not sure what to do next, my version of CUPC is 7.0(2.13496) I havem't looked for bugs yet. What version are you guys using? If anyone has any ideas please let me know ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol2 Lab1 6.1 - Weird MOH from flash
It looks like one of your gateway can receive correct multicast moh stream and the other not. I'd have a look at region settings between the MOH server (which is selected from MOH-initiating IP phone MRGL) and the gateways. For multicast you'd normally always want ulaw to be choosed (multicast MOH server should have g711-to-all region). HTH kobel On Sat, Jun 26, 2010 at 12:17 AM, Tam Nhu tamnhu...@gmail.com wrote: Not sure if someone has already run into this weird issue with MoH from flash. I've configured the Multicast MoH for both HQ H323 and BR1 MGCP gateways and they are working fine. Only that I am running into a weird issue when making outbound calls from any HQ phones to *PSTN phone line #2 (212-394-2123)*, and press Hold on HQ phone, it gives me a 'beep' moh. *However, all calls to any other PSTN lines (911, 8632683, etc) are working fine*. Moreover, I tested from BR1 phones to this PSTN line #2 and moh working fine as well. So I don't think any issues with the PSTN phone or Multicast MoH configurations. But don't know why I got 'beep' on hold just for that line from HQ phone...just weird. I reboot the router, restarted the IP VMS App service, reset all devices..still come up with the same problem.. I first don't think it is a big deal..who care. However, I sit back and though what happen in my real lab and if the proctor test your MoH, and he accidentally hit that weird line or problem...I would have a big ZERO point for that, don't I? Any suggestions? Thanks, Tam Nhu. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol2 Lab1 6.1 - Weird MOH from flash
sorry, I lied to you - MOH server is selected from the MOH-listener's MRGL (GW is your case). The MOH initiatior only decides about the source to be used. I guess it's time to bed ;) anyway, it looks like a multicast moh codec problem. you can also choose to enable g729 for the IP media streaming service. you would get this behaviour, when g729 multicast moh stream is needed and IP media streaming service can't generate it. On Sat, Jun 26, 2010 at 1:02 AM, kobel findko...@gmail.com wrote: It looks like one of your gateway can receive correct multicast moh stream and the other not. I'd have a look at region settings between the MOH server (which is selected from MOH-initiating IP phone MRGL) and the gateways. For multicast you'd normally always want ulaw to be choosed (multicast MOH server should have g711-to-all region). HTH kobel ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] AAR and ANI formatting
Make sure you're not routing the call through the translation rule, which changes the ANI. In such case the expected GW transfrom rule wouldn't match. In my case I had a translation rule, which expanded 3 digit ANI to full E.164 with + for internal calls. regards On Tue, Jun 22, 2010 at 3:43 AM, Paul Dardinski pa...@marshallcomm.comwrote: I am running into this as well. I am seeing the 4D going on the egress PSTN call even though I have explicitly configured cng txform on the gw. I also get the correct ANI on directly dialed (ie. 91617863), but on the AAR forced call () I end up with only the 4D in the cng and redir cng. Does AAR require it to be done on an explicit AAR only RP/RL ? Also, can someone clarify AAR CSS/group at the device level? Normally it is required to enable AAR at the Line level for the E164 completion (using the external phone# mask) , so is the device level AAR configuration for separate call routing? Thanks in advance, Paul (#16842 RS/Sec) * * *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *kobel *Sent:* Sunday, June 20, 2010 11:02 AM *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] AAR and ANI formatting Hello, I'm playing with AAR and VM. When out of bandwidth condition occurs, AAR correctly kicks in for different types of calls (between HQ and BR1, direct calls to VM from BR1, for incoming PSTN calls to BR1 forwarded to voicemail in HQ). It seems that the configuration is ok. But I've an issue with ANI format sent to PSTN when AAR is used. All ANI/DNIS manipulation is done on BR1 gateway via Calling/Called Party Transformation Rules. When I make a call from BR1 to VM in HQ via PSTN (explicitly, using 9.12123945600), the ANI is formatted correctly (6178631xxx/subscriber). But when I press the messages button in BR1, I can see following output from debug isdn q931 on BR1 router (outgoing SETUP): Calling Party Number i = 0x0081, '1002' Plan:Unknown, Type:Unknown Called Party Number i = 0xA0, '12123945600' Plan:Unknown, Type:National Redirecting Number i = 0x81, '5600' Plan:Unknown, Type:Unknown Surprisingly, in the CUC port monitor I can see completely different information - please compare with attached screenshot. I needed to configure alternative extension in CUC to correctly recognize the caller as CUC subscriber. It seems that AAR can handle such call correctly, but it doesn't respect the ANI transformation rules on the GW. Have you also observed this behaviour? Is there any workaround? regards kobel ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUPS presence integration with CUCM - PUBLISH vs. SUBSCRIBE
Hi, I'm not aware of any document describing this explicitly. This is the only document I know: http://docwiki.cisco.com/wiki/Cisco_Unified_Presence%2C_Release_7.x_--_How_to_Configure_the_SIP_Trunk_on_Cisco_Unified_Communications_Manager#How_to_Configure_the_SIP_Trunk_on_Cisco_Unified_Communications_Manager For me it looks strange, like it's not edited very well (in fact all steps for PUBLISH method are there, just not very clearly described and the headings seem to be incorrect) This is how I did it two times: * usual CUPS initialization wizard * create SIP trunk on CUCM with CUPS IP@ (but no other settings required) * in CUPS presence settings select this trunk (the PUBLISH checkbox is checked by default AFAIK) - this should also change CCM service parameter (PUBLISH trunk to CUPS) * create users and associate them with line appearances * configure IPPM or CUPC I didin't configure SIP trunk security profile, SUBSCRIBE CSS in CUCM, nor presence gateway in CUPS. I did it two times just to make sure that it works. I hope I'm not missing anything. After following those steps, I'm able to see presence in IPPM an CUPS - if you lift the handset on user's line, you see Busy status in IPPM/CUPC. In CCM traces there are also PUBLISH messages visible (sent from CUCM to CUPS). It would be great if somebody could confirm this independently. regards kobel 2010/6/20 Mouhammad Nasser engnasse...@hotmail.com Hi Kobel, Is there any document for how to configure the CUP Publish trunk method? I could understand from the posts that we still need to create a SIP trunk, configure it in the CUP Publish service parameter field, and assign each user to a line appearence Anyway, if the CUCM gateway is not configured, then how to tell the CUP to listen to publish information sent by CUCM on the trunk? It is really strange how Cisco left this undocumented!!! Thank you in advance Regards, ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUPS presence integration with CUCM - PUBLISH vs. SUBSCRIBE
I missed one question of yours - CUCM PUBLISH trunk is also configured in CUPS (presence - settings). There you can check the checkbox to enable PUBLISH method and select on of the SIP trunk on CUCM which should be used for this purpose. This also changes the CCM service paramter for CCM via AXL. This is how CUPS knows that it should listen to PUBLISH messages. In traces I was able to see PUBLISH messages being sent by CUCM and answered with 200 OK (CSeq: PUBLISH) by CUPS. 2010/6/20 Mouhammad Nasser engnasse...@hotmail.com Hi Kobel, Is there any document for how to configure the CUP Publish trunk method? I could understand from the posts that we still need to create a SIP trunk, configure it in the CUP Publish service parameter field, and assign each user to a line appearence Anyway, if the CUCM gateway is not configured, then how to tell the CUP to listen to publish information sent by CUCM on the trunk? It is really strange how Cisco left this undocumented!!! Thank you in advance Regards, -- Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now. https://signup.live.com/signup.aspx?id=60969 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] AAR and ANI formatting
Hello, I'm playing with AAR and VM. When out of bandwidth condition occurs, AAR correctly kicks in for different types of calls (between HQ and BR1, direct calls to VM from BR1, for incoming PSTN calls to BR1 forwarded to voicemail in HQ). It seems that the configuration is ok. But I've an issue with ANI format sent to PSTN when AAR is used. All ANI/DNIS manipulation is done on BR1 gateway via Calling/Called Party Transformation Rules. When I make a call from BR1 to VM in HQ via PSTN (explicitly, using 9.12123945600), the ANI is formatted correctly (6178631xxx/subscriber). But when I press the messages button in BR1, I can see following output from debug isdn q931 on BR1 router (outgoing SETUP): Calling Party Number i = 0x0081, '1002' Plan:Unknown, Type:Unknown Called Party Number i = 0xA0, '12123945600' Plan:Unknown, Type:National Redirecting Number i = 0x81, '5600' Plan:Unknown, Type:Unknown Surprisingly, in the CUC port monitor I can see completely different information - please compare with attached screenshot. I needed to configure alternative extension in CUC to correctly recognize the caller as CUC subscriber. It seems that AAR can handle such call correctly, but it doesn't respect the ANI transformation rules on the GW. Have you also observed this behaviour? Is there any workaround? regards kobel attachment: ScreenShot190.png___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] RE : AAR and ANI formatting
Thanks, but I forgot to mention - all the gateways are MGCP. I need to solve this on CUCM. On Sun, Jun 20, 2010 at 9:47 PM, naoufal.kerboute naoufal.kerbo...@cbi.mawrote: Try to do apply a translation-rule on the dial peer routing call to UC using ur internal extension. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UnityConnection to CUCM (Sip integration)
Hi, On SIP trunk, make sure that Diversion information is passed (redirecting number). Check also if the redirecting number is in the format expected by CUC (e.g. 4 digits only?) If not, adjust the VM profile or add alternate extension for the user in CUC. regards On Sat, Jun 19, 2010 at 5:17 PM, naoufal.kerboute naoufal.kerbo...@cbi.mawrote: Hi, I'm working on sip integration between CUCM and UnityConnection and I'm having a small problem. When the call forwarded to the a user voicemail, UC ask me for the password and not redirect call to the user mailbox. Any Idea? Regards ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] RE : RE : CBarge Not Working (Lab7 Vol2)
hi, all devices which are expected to take part in the conference should see the CFB in their MRGL. if that's not the cause of your issue, then enable troubleshooting traces for CCM service and check. file tail activelog /cm/trace/ccm/sdi recent on the processing node (subscriber) to see why you get busy tone. regards On Sat, Jun 19, 2010 at 6:54 PM, naoufal.kerboute naoufal.kerbo...@cbi.mawrote: Any Idea guys Message d'origine De: ccie_voice-boun...@onlinestudylist.com de la part de naoufal.kerboute Date: sam. 6/19/2010 2:03 À: Graham Hopkins Cc: ccie_voice@onlinestudylist.com Objet : [OSL | CCIE_Voice] RE : CBarge Not Working (Lab7 Vol2) I'v assigned only the BR2 phones to the mrgl, because I want to use the cbarge function only on bR2phon2 Message d'origine De: Graham Hopkins [mailto:ghopk...@wolf-rock.co.ukghopk...@wolf-rock.co.uk ] Date: sam. 6/19/2010 1:58 À: naoufal.kerboute Cc: ccie_voice@onlinestudylist.com Objet : Re: [OSL | CCIE_Voice] CBarge Not Working (Lab7 Vol2) Do all devices have MRGLs that can see the bridge ? Also check privacy settings but looks like they are OK if remote in use shows uo Graham On 19 Jun 2010, at 14:51, naoufal.kerboute naoufal.kerbo...@cbi.ma wrote: Hi, I'm working on lab7 Vol2 section DISA dialing, And I can't get the cbarge to work. I've configured the single button Cbarge under the BR2Phone2, also the HW conf bridge on the BR2 GW registred to the CUCM, but when I call the HQ or BR1 phones from the BR2 Mobile Phones and answer the call, I can see on BR2Phone2 that is in remote in use but when I press the line button the phone display to conference but I here a busy tone. Any Idea? Thank you guys ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUPS presence integration with CUCM - PUBLISH vs. SUBSCRIBE
I undestand, that these are two methods to achieve the same. But PUBLISH is recommended by Cisco for integrations with CUCM 7.x and higher because of performance reasons. Additionally, it's less cumbersome to configure (no Subscribe CSS needed, no SIP security profile modification) - that's an added value for us ;) Do you see any drawbacks of using PUBLISH in the lab instead of SUBSCRIBE? On Fri, Jun 18, 2010 at 1:27 PM, Moataz Mamdouh moataz_m...@yahoo.comwrote: Dear Kobel publish and subscribe SIP methods can do the same job . SUBSCRIBE event is used to order the maintain the dialog behavior ( as SUBSCRIBE is difiend as an dialog-creation method !:) ( i read it in the RFC 3265 http://www.networksorcery.com/enp/rfc/rfc3265.txt this behavior modified by a method PUBLISH that does not look at the this expiry header for each Request-URI http://www.networksorcery.com/enp/rfc/rfc3903.txt Best Regards Moataz Mamdouh CCIE # 26129 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 1 Lab 5C Q12 Mobile Voice Access
Hi, Are you using additional H.323 gateway hairpinning in order to enable MVA (i.e. your main gateway uses MGCP)? In such case this is expected bahaviour (always ask for ID). see: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/6_0_1/ccmfeat/fsmobmgr.pdf (Configuring a H.323 Gateway for System Remote Access Using Hairpinning) regards kobel On Fri, Jun 18, 2010 at 5:36 PM, Jones, Brett brett.jo...@redstone.co.ukwrote: Hi Brain, I have set my mobile number to be 2123942123 and on the debug I can see the same number coming into the gateway but still no joy. Any other ideas? Thanks Brett *From:* bkvalent...@gmail.com [mailto:bkvalent...@gmail.com] *Sent:* 18 June 2010 00:33 *To:* Jones, Brett; ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] Vol 1 Lab 5C Q12 Mobile Voice Access Yes. On your remote destination, don't put any leading digits. Don't put the 9 trunk access code. Leave it the same as your mobile ani. Use application dial rules to prefix the trunk access code. Mobile connect uses application dial rules to xform the redirecting number. MVA doesn't like it when the ANI of the caller is shorter than the configured remote destination. Brian - Reply message - From: Jones, Brett brett.jo...@redstone.co.uk Date: Thu, Jun 17, 2010 7:22 pm Subject: [OSL | CCIE_Voice] Vol 1 Lab 5C Q12 Mobile Voice Access To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Hi, I have configure mobile voice access as describe in the video walk through, however when I dial in from my mobile or any other number the IVR asks for me to enter my remote destination number followed by the pound key and not my pin number. When I enter 12345# (which is the pin number) it tell me that it's not a recognised remote destination number. I have set the service parameter to partial match and even changed the matched digits to 7. Anyone see this before? Thanks DISCLAIMER: This correspondence may contain information which is confidential or proprietary or both. Any dissemination, distribution, copying or use of this communication without prior permission of the sender is strictly prohibited. If you are not the intended recipient you may not disclose, copy or use this information. If you have received this message in error, please contact the sender to discuss its return or destruction. The contents, comments and views contained or expressed within this correspondence do not necessarily reflect those of Redstone, its subsidiaries, affiliates, associates or sister companies and are not intended to create legal relations with the recipient. Redstone may monitor email traffic data and also the content of email for the purposes of security and staff training. If you would like to know more about Redstone Converged Solutions, visit us on the web at www.redstoneconverged.co.uk or contact our Head Office on 0845 20 1. Redstone Converged Solutions Limited Registered in England Wales with Company Number: 02027207 Registered Office: Kirtlington Business Centre, Slade Farm, Kirtlington, Kidlington, Oxfordshire, OX5 3JA Click herehttps://www.mailcontrol.com/sr/3BDuQqHkNS3TndxI%21oX7UsqBPMJYdVQNfx%21p7HNYDaJwLgMpMG8nI9CV%21dDo3HBPyNfl1u4k+dgWVaQUGz3XVw==to report this email as spam. -- DISCLAIMER: This correspondence may contain information which is confidential or proprietary or both. Any dissemination, distribution, copying or use of this communication without prior permission of the sender is strictly prohibited. If you are not the intended recipient you may not disclose, copy or use this information. If you have received this message in error, please contact the sender to discuss its return or destruction. The contents, comments and views contained or expressed within this correspondence do not necessarily reflect those of Redstone, its subsidiaries, affiliates, associates or sister companies and are not intended to create legal relations with the recipient. Redstone may monitor email traffic data and also the content of email for the purposes of security and staff training. If you would like to know more about Redstone Converged Solutions, visit us on the web at www.redstoneconverged.co.uk or contact our Head Office on 0845 20 1. Redstone Converged Solutions Limited Registered in England Wales with Company Number: 02027207 Registered Office: Kirtlington Business Centre, Slade Farm, Kirtlington, Kidlington, Oxfordshire, OX5 3JA ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUPS presence integration with CUCM - PUBLISH vs. SUBSCRIBE
I also discovered today, that when using PUBLISH integration, presence gateway does not need to be configured in CUPS (which matches the description available on its configuration page in CUPS, which mentions only SUBSCRIBE messages). All this together streamlines CUPS integartion nad is more inline with Cisco recommendation. On Fri, Jun 18, 2010 at 11:13 PM, Moataz Mamdouh moataz_m...@yahoo.comwrote: I do not have any problem with the presence status , my CUPS verison is 7.0.4 i usually check the 4 fields in SIP trunk security profile and it works with me --- On *Fri, 6/18/10, kobel findko...@gmail.com* wrote: From: kobel findko...@gmail.com Subject: Re: [OSL | CCIE_Voice] CUPS presence integration with CUCM - PUBLISH vs. SUBSCRIBE To: Moataz Mamdouh moataz_m...@yahoo.com Cc: ccie_voice@onlinestudylist.com Date: Friday, June 18, 2010, 8:14 AM I undestand, that these are two methods to achieve the same. But PUBLISH is recommended by Cisco for integrations with CUCM 7.x and higher because of performance reasons. Additionally, it's less cumbersome to configure (no Subscribe CSS needed, no SIP security profile modification) - that's an added value for us ;) Do you see any drawbacks of using PUBLISH in the lab instead of SUBSCRIBE? On Fri, Jun 18, 2010 at 1:27 PM, Moataz Mamdouh moataz_m...@yahoo.comhttp://us.mc559.mail.yahoo.com/mc/compose?to=moataz_m...@yahoo.com wrote: Dear Kobel publish and subscribe SIP methods can do the same job . SUBSCRIBE event is used to order the maintain the dialog behavior ( as SUBSCRIBE is difiend as an dialog-creation method !:) ( i read it in the RFC 3265 http://www.networksorcery.com/enp/rfc/rfc3265.txt this behavior modified by a method PUBLISH that does not look at the this expiry header for each Request-URI http://www.networksorcery.com/enp/rfc/rfc3903.txt Best Regards Moataz Mamdouh CCIE # 26129 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 1 Lab 5C Q12 Mobile Voice Access
Thanks, I guess I'll need to check this. Was there some limitation with such workaround? I remind myself seeing something like this in the proctor guide, but can't remember exactly now. thanks, kobel On Fri, Jun 18, 2010 at 6:37 PM, Angel Perez gorr...@hotmail.com wrote: Hi, you can avoid this behaviour configuring a voice translation rule at h323 gw at incoming dial-peer from ucm route pattern. Transform calling number to rd number. Also you can do this at route pattern with a mask. hth -- Date: Fri, 18 Jun 2010 18:11:06 +0200 From: findko...@gmail.com To: brett.jo...@redstone.co.uk CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 5C Q12 Mobile Voice Access Hi, Are you using additional H.323 gateway hairpinning in order to enable MVA (i.e. your main gateway uses MGCP)? In such case this is expected bahaviour (always ask for ID). see: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/6_0_1/ccmfeat/fsmobmgr.pdf (Configuring a H.323 Gateway for System Remote Access Using Hairpinning) regards kobel On Fri, Jun 18, 2010 at 5:36 PM, Jones, Brett brett.jo...@redstone.co.ukwrote: Hi Brain, I have set my mobile number to be 2123942123 and on the debug I can see the same number coming into the gateway but still no joy. Any other ideas? Thanks Brett *From:* bkvalent...@gmail.com [mailto:bkvalent...@gmail.com] *Sent:* 18 June 2010 00:33 *To:* Jones, Brett; ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] Vol 1 Lab 5C Q12 Mobile Voice Access Yes. On your remote destination, don't put any leading digits. Don't put the 9 trunk access code. Leave it the same as your mobile ani. Use application dial rules to prefix the trunk access code. Mobile connect uses application dial rules to xform the redirecting number. MVA doesn't like it when the ANI of the caller is shorter than the configured remote destination. Brian - Reply message - From: Jones, Brett brett.jo...@redstone.co.uk Date: Thu, Jun 17, 2010 7:22 pm Subject: [OSL | CCIE_Voice] Vol 1 Lab 5C Q12 Mobile Voice Access To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Hi, I have configure mobile voice access as describe in the video walk through, however when I dial in from my mobile or any other number the IVR asks for me to enter my remote destination number followed by the pound key and not my pin number. When I enter 12345# (which is the pin number) it tell me that it's not a recognised remote destination number. I have set the service parameter to partial match and even changed the matched digits to 7. Anyone see this before? Thanks DISCLAIMER: This correspondence may contain information which is confidential or proprietary or both. Any dissemination, distribution, copying or use of this communication without prior permission of the sender is strictly prohibited. If you are not the intended recipient you may not disclose, copy or use this information. If you have received this message in error, please contact the sender to discuss its return or destruction. The contents, comments and views contained or expressed within this correspondence do not necessarily reflect those of Redstone, its subsidiaries, affiliates, associates or sister companies and are not intended to create legal relations with the recipient. Redstone may monitor email traffic data and also the content of email for the purposes of security and staff training. If you would like to know more about Redstone Converged Solutions, visit us on the web at www.redstoneconverged.co.uk or contact our Head Office on 0845 20 1. Redstone Converged Solutions Limited Registered in England Wales with Company Number: 02027207 Registered Office: Kirtlington Business Centre, Slade Farm, Kirtlington, Kidlington, Oxfordshire, OX5 3JA Click herehttps://www.mailcontrol.com/sr/3BDuQqHkNS3TndxI%21oX7UsqBPMJYdVQNfx%21p7HNYDaJwLgMpMG8nI9CV%21dDo3HBPyNfl1u4k+dgWVaQUGz3XVw==to report this email as spam. -- DISCLAIMER: This correspondence may contain information which is confidential or proprietary or both. Any dissemination, distribution, copying or use of this communication without prior permission of the sender is strictly prohibited. If you are not the intended recipient you may not disclose, copy or use this information. If you have received this message in error, please contact the sender to discuss its return or destruction. The contents, comments and views contained or expressed within this correspondence do not necessarily reflect those of Redstone, its subsidiaries, affiliates, associates or sister companies and are not intended to create legal relations with the recipient. Redstone may monitor email traffic data and also the content of email for the purposes of security and staff training. If you would
Re: [OSL | CCIE_Voice] RTP Flow through gatekeepe
you should be able to see the RTP flows set up by CUBE via show voip rtp connections command. regards kobel On Thu, Jun 17, 2010 at 6:53 PM, Checker CCIEV cciev2...@gmail.com wrote: Thanks amy for your inputs Once call established is there any specific show or debug command to verify that rtp flow through the gatekeeper. Anyway I will try this configuration..give you feed back On Thu, Jun 17, 2010 at 8:13 PM, Amy Ryan ar...@ipexpert.com wrote: Yes, you could do this as an IP2IP GW (Gatekeeper CUBE) Here is an example configuration for the gatekeeper itself: gatekeeper zone local SPAIN ipexpert.com 10.10.110.1 outvia CUBE zone local US ipexpert.com outvia CUBE zone local CUBE ipexpert.com zone prefix SPAIN 3* zone prefix US 5* gw-type-prefix 1#* default-technology no shutdown Of course you will need to register the endpoints add the appropriate rp’s, dialpeers and such to properly route the calls as desired. :-) HTH, Amy --- Amy Ryan – CCIE #24677 (Voice) Technical Instructor - IPexpert, Inc. Mailto: *ar...@ipexpert.com *Telephone: +1.810.326.1444 Live Assistance, Please visit: www.ipexpert.com/chat * http://www.ipexpert.com/chat* eFax: +1.810.454.0130 IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities *http://www.ipexpert.com/communities* and our public website at www.ipexpert.com *http://www.ipexpert.com/* -- *From: *Checker CCIEV cciev2...@gmail.com *Date: *Thu, 17 Jun 2010 19:55:45 +0400 *To: *ccie_voice@onlinestudylist.com *Subject: *[OSL | CCIE_Voice] RTP Flow through gatekeepe How can we get rtp flow through gatekeeper? Normally after call set up rtp flow directly between end points. is it achievable using ip2ip gw or gatekeeper proxy (use-proxy command) -- ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CUPS presence integration with CUCM - PUBLISH vs. SUBSCRIBE
Dear fellow CCIE wannabies, I was wondering what are your experiences with integrating CUPS with CUCM. AFAIK, there are two methods: * SUBSCRIBE - has to be used with CUCM releases = 6.x (requires SIP trunk with correct SIP sec profile) * PUBLISH - can be used with 7.x and later (PUBLISH trunk needs to be configured in CCM service params and in CUPS) The second one is recommended for performance reasons. The difference is that with SUBSCRIBE, CUPS needs to sent to a CUCM a SUBSCRIBE message for specific presentity (e.g. DN) and only then receives NOTIFY messages with the presence state. With PUBLISH method, the CUCM sends PUBLISH messages to CUPS without any need for previous SUBSCRIBE messages. Now the question - proctor guide and CUPS documentation recommend configuring a SIP trunk with SIP Security Profile which allow SUBSCRIBE messages and SUBSCRIBE CSS. This would point to the first mechanism being used. But it seems that I'm able to have working presence without this, only configuring PUBLISH trunk (in CCM service parameters and in CUPS itself) - I see the line presence information in CUPC and IPPM - this shows that PUBLISH method works. I was wondering if you have the same experiences? Am I missing something about the way PUBLISH works? It looks like quicker configurtion method with the same results. Does it have any drawbacks I can't see? regards kobel ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUPS presence integration with CUCM - PUBLISH vs. SUBSCRIBE
One more thing - I know I wrote Cisco documentation recommends, but we all know the CUPS documentation... For me it looks like it has not been updated after PUBLISH method became available with 7.x. In one place there is information that PUBLISH is supported/recommended with CUCM 7.x, but on the wiki page describing CUCM configuration, there is only information how to configure SUBSCRIBE methods. On Thu, Jun 17, 2010 at 8:54 PM, kobel findko...@gmail.com wrote: Dear fellow CCIE wannabies, I was wondering what are your experiences with integrating CUPS with CUCM. AFAIK, there are two methods: * SUBSCRIBE - has to be used with CUCM releases = 6.x (requires SIP trunk with correct SIP sec profile) * PUBLISH - can be used with 7.x and later (PUBLISH trunk needs to be configured in CCM service params and in CUPS) The second one is recommended for performance reasons. The difference is that with SUBSCRIBE, CUPS needs to sent to a CUCM a SUBSCRIBE message for specific presentity (e.g. DN) and only then receives NOTIFY messages with the presence state. With PUBLISH method, the CUCM sends PUBLISH messages to CUPS without any need for previous SUBSCRIBE messages. Now the question - proctor guide and CUPS documentation recommend configuring a SIP trunk with SIP Security Profile which allow SUBSCRIBE messages and SUBSCRIBE CSS. This would point to the first mechanism being used. But it seems that I'm able to have working presence without this, only configuring PUBLISH trunk (in CCM service parameters and in CUPS itself) - I see the line presence information in CUPC and IPPM - this shows that PUBLISH method works. I was wondering if you have the same experiences? Am I missing something about the way PUBLISH works? It looks like quicker configurtion method with the same results. Does it have any drawbacks I can't see? regards kobel ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUPS presence integration with CUCM - PUBLISH vs. SUBSCRIBE
...at least one person tends to agree ;) http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg09296.html On Thu, Jun 17, 2010 at 9:02 PM, kobel findko...@gmail.com wrote: One more thing - I know I wrote Cisco documentation recommends, but we all know the CUPS documentation... For me it looks like it has not been updated after PUBLISH method became available with 7.x. In one place there is information that PUBLISH is supported/recommended with CUCM 7.x, but on the wiki page describing CUCM configuration, there is only information how to configure SUBSCRIBE methods. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] QOS FRF.12 MLPP
Also, after the Auto QOS generates a lot of classes etc. We do edit few things here and there. Just wanted to confirm that is it a good practice to remove rtp header compression? I use to remove it always but now I am getting conflicting feedback that should we remove it or not? interface Serial0/2/0.1 point-to-point bandwidth 256 frame-relay interface-dlci 301 CISCO class AutoQoS-FR-Se0/2/0-301 auto qos voip trust * frame-relay ip rtp header-compression* I would appreciate any input in this regard. you can configure cRTP in two ways. if the task doesn't explicitly ask for CB cRTP, I keep auto qos config - why waste time? I'm not aware of any drawback of this method. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Real time tracing of IP phone activity
Just a side note - I also use those two commands, but with a little modification: file tail activelog /cm/trace/cmi/sdi recent this makes the CLI to choose the most recent file (no need to type in the filename yourself). RTMT is such a waste of time, when it comes to traces ;) BTW, the most useful are SDI traces (SDLs are less readable and are used for inter-ccm communications - I never use them in lab). It's easy to remember - SD-III like IIIncredibly useful traces :D regards kobel On Tue, Jun 15, 2010 at 1:00 PM, Matthew Berry ciscovoiceg...@gmail.comwrote: I would turn on detailed tracing through CUCM Serviceability and then monitoring the SDL or SDI traces (I always forget which one) through the CUCM CLI. It's the best way I know how. file tail activelog /cm/trace/cmi/sdl file tail activelog /cm/trace/cmi/sdi ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] service redundancy
AFAIK, there is also another solution - round robin resolving on DNS server. but it's also out of the scope probably. I can't point to any document right now, but AFAIR to configure phone services for both EM on Pub and Sub and let user manually select the working one. On Tue, Jun 15, 2010 at 3:24 PM, wolfsrudel wolfsru...@gmail.com wrote: you can gen em redundancy by means of slb in the gateway, but it's out of the scope of the current blueprint. don't have any link right know but can be looked up easily @ cisco. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Spanning-tree portfast IPexpert!
AFAIK, the portfast is effective only for access ports. that's why it's not visible on HWIC-4ESW ports, which are configured in trunk mode. regards On Sun, Jun 13, 2010 at 6:49 PM, wolfsrudel wolfsru...@gmail.com wrote: portfast should be set on any access port where we like to avoid stp delays (learning and such). it's part of the de facto port config configuracion, unless were have specific reasons not to do so. imho hth On 6/13/10, Ashar Siddiqui siddas...@gmail.com wrote: Hi, In Proctor lab HW-Switch I can see this command: interface FastEthernet1/0/2 switchport access vlan 10 switchport mode access switchport voice vlan 20 spanning-tree portfast But Spanning-tree portfast is not used on BR1/BR2 ports where phones are connected. Any specific reason? I thought we will use this command anywhere where we want the ports not to come in Election process of Root bridge (STP) and we are sure that they won't create ant loops (like access ports or ports connected to phone). Also they quickly go in forwarding state..Why are we not using this on Br1 and Br2? Ash -- Sent from my mobile device ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Is there any way to restart (not reset) the phone from phone level?
Hi, I sometimes have a strange issue with CUCM - when I click Restart button from CUCM administration the phone restarts only after substantial delay (1-2 minutes) - this way configuration changes take ages to complete. I know I can reset the phone with **#**, but it takes a lot of time and is not always required and restart is much quicker. Is there any way to perform only restart (that is only re-registration, not reset and config download)? Generally, on what occasions you perform phone reset and not restart (i.e. restart doesn't apply the changes)? regards kobel ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] H323 gateway inbound calls issue.
http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg17030.html hth On Sat, Jun 12, 2010 at 7:26 PM, Paul Smith psm...@netcraftsmen.net wrote: A thought off the top of MY head... Regarding the following configuration: interface vlan 302 h323-gateway voip bind srcaddr 1.1.1.1 I've always found that things work better when you have h323-gateway voip interface on the interface as well. Paul ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCCX reporting
OK, this might work. But another part of the task is providing this information to the customer. The task is vague about this - how would you do this? 2010/6/11 Mouhammad Nasser engnasse...@hotmail.com Hi Kobel, In the Get Reporting Statistic step, one can retrieve the number of total contacs in queue, which is defined as: Number of total contacts since the statistics were last reset for this CSQ But I didn't try it before, so I am not sure if one can choose which CSQ to retrieve information about, or we have to: - apply select resource (with no connect) - Retrieve the required statistics - dequeue the call from that queue and repeat the above till all informatoin is available HTH Mouhammad, -- Hotmail: Powerful Free email with security by Microsoft. Get it now.https://signup.live.com/signup.aspx?id=60969 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCCX reporting
Thanks, I had the same idea. But the generality of this task made me think if I'm not missing any built-in feature. 2010/6/11 Mouhammad Nasser engnasse...@hotmail.com Hi, Well, I cannot think of something other than recording customized prompts like: The choice with the highest number of clicks is, followed by another prompt saying the name of the queue. We need if statements here to choose the correct CSQ to mention the name of ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCCX CSQ hunting order issue
I would expect this - it's not a rocket science. Possibly another issue thanks to VMware - I was wondering if anyone seen this also. thanks, kobel On Thu, Jun 10, 2010 at 4:52 AM, wolfsrudel wolfsru...@gmail.com wrote: i've tested this today and works fine, all call are first delivered to the first agent. On Wed, Jun 9, 2010 at 5:32 PM, wolfsrudel wolfsru...@gmail.com wrote: easiest would be routing by skill (most skilled). if one of the agents has a higher weight (on that skill, not the weight attribute) then any call should always be delivered to the same agent always, no matter what. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCCX CSQ hunting order issue
VMWare ESXi 4.0.0 UCCX 7.0(1)_Build168 On Thu, Jun 10, 2010 at 10:33 AM, Angel Perez gorr...@hotmail.com wrote: Hi: Wich vmware version do you have installed? I'm working with esxi and I've never seen this ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] vol 2 lab 6: UCCX reporting
Hello, In vol 2 lab 6 UCCX task 10.2 there is a requirement to provide reporting based on options choosen by callers. I.e. a caller is presented with a menu of options (press 0 for operator, press 1 for directory, etc) and may choose a digit to proceed - the customer should be able to check which branch is choosen most frequently. The proctor guide seems to ignore this requirement. Any idea how this could be achieved? regards kobel ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] UCCX CSQ hunting order issue
Hi, I'm trying to configure CSQ in such way, that incoming calls are always reouted to the first agent, then to the other (vol 2 lab 10) I see 2 ways of achieving this: * configure CSQ with resource group selection model + resource selection criteria set to Linear * configure CSQ with skill based routing, assign appropriate skills to the agents and choose Most/Least skilled selection criteria However, none of these work. I always get the same behaviour: circular call routing (each new call ends up at different agent) Am I missing something? I even tried restarting the UCCX engine. UCCX is on a VMWare. I had a nasty bug with it - couldn't configure one button login. After few hours of fighting with it I on VMWare (didn't succeed) I configured it in 3 minutes on another UCCX installation on a physical server. what a waste of time... It makes me wonder, if this is not something similar. best regards kobel ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Connecting SIP phone to Proctor Lab BR2
in your case it might be an issue with firmware authentication (was the phone registered earlier to a newer CUCM?) the procedure to register any phone in CME: * perform factory reset of the phone * debug tftp server events - check what files it's looking for (default loads) * provide those files with tftp-server comand On Mon, Jun 7, 2010 at 2:05 AM, Steve Sarrick ssarr...@drsllc.net wrote: I have a 7962 that I am battling and I am giving in for a little direction! I am working on Lab 5 Vol 1 for example. I am trying to register my local 7962 as Phone 4 which is the SIP phone on BR2 (CME). All other phones are up fine. My phone begins the upgrading process and fails after downloading the dsp file with an Auth Fail message. I have all the tftp-server lines in for the 7962 (I am fairly sure). I am at a bit of a loss here as I have tried several things. I am using the files that are on BR2. Do I have to use a different version (I would think that PL has appropriate versions). Am I missing something configuration wise. Anyone else with a similar issue or a direction to try. Part of me is thinking that PL would have everything ready to go, so I don’t want to stray too far in terms of going outside of what they have provided. It tends to make me think I am missing something obvious but for as little as I have worked with CME SIP Phones, I don’t want to second guess. Any help is appreciated. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Gatekeeper 4 Digit call
try configuring higher preference for CUCM trunk for prefix 2* . It seems that GK performs load balancing between the CUBE and this trunk and only one of them works. regards On Mon, Jun 7, 2010 at 10:36 PM, Dani Bug daniyal.vo...@gmail.com wrote: I tried without invia/outvia still no luck ...:( HQ-R1#sh gatekeeper gw GATEWAY TYPE PREFIX TABLE = Prefix: 1* Zone GK master gateway list: 142.102.64.254:1720 CUBE 172.25.105.101:1720 GK_Trunk_1 Prefix: 852* Zone GK master gateway list: 142.102.66.254:1720 CUCME Here is debug success call from 4001 to 2001 HQ-R1# Jun 7 22:07:34.108: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Jun 7 22:07:34.172: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Jun 7 22:07:34.172: ////GK/gk_rassrv_arq: arqp=0x49E870B4,crv=0x59, answerCall=0 Jun 7 22:07:34.172: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC Jun 7 22:07:34.172: //E801AC8F8482/E80248B78484/GK/gk_dns_query: No Name servers Jun 7 22:07:34.172: //E801AC8F8482/E80248B78484/GK/rassrv_get_addrinfo: (12001) Matched tech-prefix 1 Jun 7 22:07:34.176: //E801AC8F8482/E80248B78484/GK/rassrv_get_addrinfo: (12001) Matched zone prefix 2 and remainder 001 Jun 7 22:07:34.176: ////GK/gk_rassrv_get_ingress_network: ARQ non-std ingress network = 1 Jun 7 22:07:34.176: //E801AC8F8482/E80248B78484/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x49FFA4B8 Jun 7 22:07:34.176: //E801AC8F8482/E80248B78 HQ-R1#484/GK/rassrv_arq_select_viazone: matched zone is GK, and z_invianamelen=2 Jun 7 22:07:34.176: //E801AC8F8482/E80248B78484/GK/rassrv_arq_select_viazone and z_invianamep=GK Jun 7 22:07:34.176: //E801AC8F8482/E80248B78484/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x49FFA4B8 Jun 7 22:07:34.176: //E801AC8F8482/E80248B78484/GK/rassrv_arq_select_viazone: matched zone is GK, and z_outvianamelen=2 Jun 7 22:07:34.176: //E801AC8F8482/E80248B78484/GK/rassrv_arq_select_viazone and z_outvianamep=GK Jun 7 22:07:34.176: //E801AC8F8482/E80248B78484/GK/rassrv_arq_select_viazone: Received ARQ for a zone (GK) that has an outviazone (GK) specified, but I am that viazone. Continue normal ARQ processing Jun 7 22:07:34.176: ////GK/gk_rassrv_get_ingress_network: ARQ non-std ingress network = 1 Jun 7 22:07:34.192: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Jun 7 22:07:34.192: ////GK/gk_rassrv_arq: arqp=0x49E870B4,crv=0x8059, answerCall=1 Jun 7 22:07:34.192: //E801AC8F8482/E80248B78484/GK/gk_rassrv_dep_arq: ARQ Didn't use GK_AAA_PROC Jun 7 22:07:35.268: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Jun 7 22:07:35.268: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup HQ-R1# = Here is debug for Failed call from 4001 to 2001 HQ-R1# Jun 7 22:07:44.888: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Jun 7 22:07:44.892: ////GK/gk_rassrv_arq: arqp=0x49EB9F60,crv=0x5B, answerCall=0 Jun 7 22:07:44.892: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC Jun 7 22:07:44.892: //EE6A58F08490/EE6A58F08492/GK/gk_dns_query: No Name servers Jun 7 22:07:44.892: //EE6A58F08490/EE6A58F08492/GK/rassrv_get_addrinfo: (12001) Matched tech-prefix 1 Jun 7 22:07:44.892: //EE6A58F08490/EE6A58F08492/GK/rassrv_get_addrinfo: (12001) Matched zone prefix 2 and remainder 001 Jun 7 22:07:44.892: ////GK/gk_rassrv_get_ingress_network: ARQ non-std ingress network = 1 Jun 7 22:07:44.892: //EE6A58F08490/EE6A58F08492/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x49FFA4B8 Jun 7 22:07:44.892: //EE6A58F08490/EE6A58F08492/GK/rassrv_arq_select_viazone: matched zone is GK, and z_invianamelen=2 Jun 7 22:07:44.89 HQ-R1#2: //EE6A58F08490/EE6A58F08492/GK/rassrv_arq_select_viazone and z_invianamep=GK Jun 7 22:07:44.892: //EE6A58F08490/EE6A58F08492/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x49FFA4B8 Jun 7 22:07:44.892: //EE6A58F08490/EE6A58F08492/GK/rassrv_arq_select_viazone: matched zone is GK, and z_outvianamelen=2 Jun 7 22:07:44.892: //EE6A58F08490/EE6A58F08492/GK/rassrv_arq_select_viazone and z_outvianamep=GK Jun 7 22:07:44.892: //EE6A58F08490/EE6A58F08492/GK/rassrv_arq_select_viazone: Received ARQ for a zone (GK) that has an outviazone (GK) specified, but I am that viazone. Continue normal ARQ processing Jun 7 22:07:44.892: ////GK/gk_rassrv_get_ingress_network: ARQ non-std ingress network = 1 Jun 7 22:07:44.912: ////GK/gk_process: QUEUE_EVENT (minor 0)
Re: [OSL | CCIE_Voice] Lab 1 Volume 2 questions
hi, 1 - you were supposed to have 1 leg in g711 and 1 in g729 with xcode between them on CUBE 2 - g729r8 is not what you want. it's higher complexity then g729a, and only g729a is used by Cisco phones. 3 - it worked the same for me AFAIR. for each new call, the other phone rung. I didn't create any AC users - AFAIK, they are used only when desktop app is used, which is not a case now (not supported anymore with new installations). I assumed its expected behaviour, but I'm curious if anybody has different opinion on this ;) On Mon, Jun 7, 2010 at 10:41 PM, Daniel Berlinski dberlin...@gmail.comwrote: Hello all Hope you all doing well. I would like to bring to you guys attention and hopefully get some interesting replies on some technology topics of Lab 1 of Volume 2 that I am not 100% sure about. 1- Gatekeeper section: Had a problem with the calls between CME and CUCM taking all WAN bandwidth overtime. This was solved after completing the CAC section by issuing “bandwidth zone UCM 32” command in gatekeeper. That being said a couple of things come to mind: Show gatekeeper calls does not show the same output as asked back in sections 4.2 and 4.3 of the lab, secondly PG suggests that the CAC sestion could also be solved by issuing a gatekeeper command for the CME zone but that would be 240Kbps. I did not understand why this was suggested as I believe we have 2 call legs here. Right? For reference this was mentioned on page 99 of the Proctor Guide. I’m particularly interested in clarifying this question because I suspect I’m missing something fundamental here. My understanding is that we are talking 2 g711ulaw call legs over the WAN between the CUBE and CME right? 2- It is not clear why it is suggested not to include g729r8 in IOS xcoder configuration as I believe this is necessary in situations where g729r8 is the codec that needs xcoding 3- Attendant Console question what is the expected behaviour while testing? I ring the pilot point and it rings only in one of the 2 extensions never hunting over to the next. How did this work for you? Have you created users and logged in to the Attendant console CTI app to get it to hunt properly? 4- What keywords in the call routing/Device Mobility section defined the requirements for configuring the US sites in the same DMG? I decided to configure those 2 device pools in different DMGs because of the question stating neet not to keep class of restriction - I based my decision in configuring not to inherit roaming sensitve settings on that statement. 7 dgt ANI presentation without name for 911 calls was preserved becaue HQ LRG was used by BR1 Phone while roaming. Was it wrong? Best regards Daniel ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 1 Volume 2 questions
what do you mean? they are the same at bit level (no way to recognize, which one was used to encode the audio), but the processing power needed to encode voice with g729r8 is higher then with g729ar8 (because g729ar8 makes some trade-off on quality) On Mon, Jun 7, 2010 at 11:42 PM, Dan C Williams dan.c.willi...@gmail.comwrote: Concerning #2: Just to clarifyg729r8 is g729a -- Dan C Williams On Tue, Jun 8, 2010 at 02:07, kobel findko...@gmail.com wrote: hi, 1 - you were supposed to have 1 leg in g711 and 1 in g729 with xcode between them on CUBE 2 - g729r8 is not what you want. it's higher complexity then g729a, and only g729a is used by Cisco phones. 3 - it worked the same for me AFAIR. for each new call, the other phone rung. I didn't create any AC users - AFAIK, they are used only when desktop app is used, which is not a case now (not supported anymore with new installations). I assumed its expected behaviour, but I'm curious if anybody has different opinion on this ;) On Mon, Jun 7, 2010 at 10:41 PM, Daniel Berlinski dberlin...@gmail.comwrote: Hello all Hope you all doing well. I would like to bring to you guys attention and hopefully get some interesting replies on some technology topics of Lab 1 of Volume 2 that I am not 100% sure about. 1- Gatekeeper section: Had a problem with the calls between CME and CUCM taking all WAN bandwidth overtime. This was solved after completing the CAC section by issuing “bandwidth zone UCM 32” command in gatekeeper. That being said a couple of things come to mind: Show gatekeeper calls does not show the same output as asked back in sections 4.2 and 4.3 of the lab, secondly PG suggests that the CAC sestion could also be solved by issuing a gatekeeper command for the CME zone but that would be 240Kbps. I did not understand why this was suggested as I believe we have 2 call legs here. Right? For reference this was mentioned on page 99 of the Proctor Guide. I’m particularly interested in clarifying this question because I suspect I’m missing something fundamental here. My understanding is that we are talking 2 g711ulaw call legs over the WAN between the CUBE and CME right? 2- It is not clear why it is suggested not to include g729r8 in IOS xcoder configuration as I believe this is necessary in situations where g729r8 is the codec that needs xcoding 3- Attendant Console question what is the expected behaviour while testing? I ring the pilot point and it rings only in one of the 2 extensions never hunting over to the next. How did this work for you? Have you created users and logged in to the Attendant console CTI app to get it to hunt properly? 4- What keywords in the call routing/Device Mobility section defined the requirements for configuring the US sites in the same DMG? I decided to configure those 2 device pools in different DMGs because of the question stating neet not to keep class of restriction - I based my decision in configuring not to inherit roaming sensitve settings on that statement. 7 dgt ANI presentation without name for 911 calls was preserved becaue HQ LRG was used by BR1 Phone while roaming. Was it wrong? Best regards Daniel ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME 7.0 Presence caller-list is not working ...
try create cnf-files restart the phones. On Sat, Jun 5, 2010 at 4:21 AM, Shadow of Voice salman.shaik...@gmail.comwrote: Hi Guys I have issue when configure presence in CME I allow subscribe and allow watch globally still can't see caller list on missed call does any one know where i am wrong and why my CME presence caller-list is not working ! presence presence call-list allow subscribe ! ephone-dn 2 octo-line number 4002 no-reg primary description +6524044002 name SiteC-Ph2 allow watch call-forward busy 4220 call-forward noan 4220 timeout 20 ! ! ephone 1 device-security-mode none mac-address 001A.A1C8.0H8F ephone-template 1 blf-speed-dial 1 4002 label SiteC-Ph2 type 7961 button 1:1 3:3 4:5 ! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME 7.0 Presence caller-list is not working ...
and maybe sip-ua presence enable will help? On Sat, Jun 5, 2010 at 12:16 PM, kobel findko...@gmail.com wrote: try create cnf-files restart the phones. On Sat, Jun 5, 2010 at 4:21 AM, Shadow of Voice salman.shaik...@gmail.com wrote: Hi Guys I have issue when configure presence in CME I allow subscribe and allow watch globally still can't see caller list on missed call does any one know where i am wrong and why my CME presence caller-list is not working ! presence presence call-list allow subscribe ! ephone-dn 2 octo-line number 4002 no-reg primary description +6524044002 name SiteC-Ph2 allow watch call-forward busy 4220 call-forward noan 4220 timeout 20 ! ! ephone 1 device-security-mode none mac-address 001A.A1C8.0H8F ephone-template 1 blf-speed-dial 1 4002 label SiteC-Ph2 type 7961 button 1:1 3:3 4:5 ! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] vol 2 lab 1: gatekeeper and H.323 GW on the same router
I consulted Cisco IOS H.323 Configuration Guide. h323-gateway voip interface: To configure a Cisco device as an H.323 gateway in a service provider environment, configure at least one of its interfaces as a gateway interface. Use either an interface that is connected to the gatekeeper or a loopback interface for the gateway interface. h323-gateway voip bind srcaddr ip-addres: H.323 support for virtual interfaces allows the IP address of the gateway to be configured so that the IP address included in the H.323 packet is always the source IP address of the gateway, regardless of the physical interface and protocol used. This single-address feature allows firewall applications to be easily configured to work with H.323 messages. As I see it: * when interface is configured with the first command, it'll listen to H.323 signaling on its address, but signaling messages sent from the GW will use it's IP address of the interface facing the destination (might be different) * the second command makes the GW always use the configured scraddr as signaling source IP address in H.323 messges. So it means, that if I configure the H.323 GW on CUCM, I need to use the IP address which is configured on GW's interface, which has h323-gateway voip interface command. But if I want the GW to register with GK with specific IP address, I need the second command. Anybody disagrees? On Sat, Jun 5, 2010 at 2:53 AM, Daniel Zeiger Berlinski dberlin...@gmail.com wrote: Hello Kobel I'm doing a few tests here and it seems that the difference between those 2 commands relates to which interface you are sourcing your RAS and H225 packets. For instance if you remove h323-gateway voip bind srcaddr from the interface which contains the IP address of the CUCM configured H323 gateway I believe you are going to break your incoming HQ local PSTN calls. Let me know if you found any different result. Thanks On Sat, Jun 5, 2010 at 3:37 AM, kobel findko...@gmail.com wrote: bingo! indeed, this solved the issue. This command obviously binds all H.323 signaling to specific interface (loopback in my case). So this explains why the incoming calls were associated with the GK-controlled trunk (GK configured on loopback). After removing this command, the source address is bound with voice interface IP address, which is ok. Another lesson learned - for calls routed via GK, the SETUP which CUCM receives contains the source signaling IP address of the GK (despite the fact it's sent directly by the remote GW). This is why CUCM needs to sent ARQ to GK. Only ACF contains the signaling IP address of the remote GW. Now it makes perfect sense, but I've never though about it. But still, one thing is not clear for me. What's the difference between: * h323-gateway voip interface * h323-gateway voip bind srcaddr 10.225.100.254 The command reference is not very clear on this. Thanks for your input! On Fri, Jun 4, 2010 at 5:08 PM, Pavan pav.c...@gmail.com wrote: in your existing config, remove the h323gw bind source interface command ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MGCP debug messages
I think you can't see this kind of information (i.e. connection status) on the gateway itself. The calls are managed by the call agent, and it's there where the full GW state is kept. On Sat, Jun 5, 2010 at 2:55 PM, jammer jones jammerjone...@gmail.comwrote: What debugs in the router can i use to see the following information communication between a mgcp gateway and mgcp call agent(subscriber) that shows the status of a connected call. And then when the primary call-agent(subscriber) fails and the mgcp gateway switches to the redundant mgcp call agent(publisher) i want to see the same communications between the mgcp gateway and the new mgcp agent(the publisher) I have tried the following command and none of them give me this information. debug ccm-manager backhaul debug ccm-manager events debug mgcp packets debug mgcp events debug mgcp - crashes the routers?\ I have used CUCM traces with RTMT and I think i am seeing this information, but not 100% sure ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] vol 2 lab 1: gatekeeper and H.323 GW on the same router
Hello, I was wondering if anybody could help me with a little issue I have with vol 2 lab 1, call routing section. There are two requirements: * configure HQ GW as H.323 gateway in CUCM (for incoming/outgoing PSTN calls), which should use g711 codec * configure HQ GW as a GK-controlled trunk on CUCM, which should use g729 I added HQ GW as H.323 gateway using HQ router's voice interface IP address and as a GK using loopback address (CUCM doesn't allow to use the same IP address for trunk and gateway). I'm able to route the outgoing calls from CUCM via both devices (i.e. GK-controlled trunk and H.323 gateway, with appropriate region setting, digit mainipulation, etc.), but the problem I have is with incoming calls. My understaning is that calls from PSTN should be routed directly to CUCM without GK involvment (session target ipv4:CUCM_IP). But it seems, that incoming calls from HQ GW always use the loopback address as signaling source IP address (independent from the fact if they are routed via GK or directly) and therefore, they hit GK-controlled trunk device on CUCM. This results in ARQ being sent by CUCM to GK, g729 codec being used, etc. This is OK for calls coming from CUBE, but it's not OK for calls coming from PSTN (which should be routed without GK). I'm out of ideas. I'm not aware of any configuration commands to bind different IP addresses for GK-controlled H.323 calls vs. P2P H.323 calls. How did you solve this? regards kobel ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] vol 2 lab 1: gatekeeper and H.323 GW on the same router
I described from CUCM perspective: incoming calls - call from GW to CUCM. On Fri, Jun 4, 2010 at 4:43 PM, Ashar Siddiqui siddas...@gmail.com wrote: You are talking about inbound calls or outbound calls from the gateway? Sorry it’s not clear for me. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] vol 2 lab 1: gatekeeper and H.323 GW on the same router
bingo! indeed, this solved the issue. This command obviously binds all H.323 signaling to specific interface (loopback in my case). So this explains why the incoming calls were associated with the GK-controlled trunk (GK configured on loopback). After removing this command, the source address is bound with voice interface IP address, which is ok. Another lesson learned - for calls routed via GK, the SETUP which CUCM receives contains the source signaling IP address of the GK (despite the fact it's sent directly by the remote GW). This is why CUCM needs to sent ARQ to GK. Only ACF contains the signaling IP address of the remote GW. Now it makes perfect sense, but I've never though about it. But still, one thing is not clear for me. What's the difference between: * h323-gateway voip interface * h323-gateway voip bind srcaddr 10.225.100.254 The command reference is not very clear on this. Thanks for your input! On Fri, Jun 4, 2010 at 5:08 PM, Pavan pav.c...@gmail.com wrote: in your existing config, remove the h323gw bind source interface command ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
