[OSL | CCIE_Voice] Calling Party Transformation
Hello Experts, How do I set up a Calling Party Transformation in DP to trap any devices that did not have any caller ID assigned and assign them to a general number? How do I assign pattern to match null caller ID in the calling party transformation? Thank you, Bo ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] MOH on FXO port
Hello experts, I have a router that included both FXO ports and digital T1 cards. I configured MoH on CUCME. When callers call into CUCME via ISDN, they could hear MoH without any problem. Callers call in from FXO could not hear anything. Does anyone know what I should do to fix the problem? Thank you, Bo ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] auto registration not work
Why would you need IP helper-address command in your vlan 400? It seems to me you already had your SC router set up as DHCP server. Bo On Wed, Apr 27, 2011 at 8:05 PM, donny f f.faraday...@gmail.com wrote: yes, cause my UCM site BR-1 and HQ register ok I did not check which state , filter message ? no i did not , it is Proctor lab, never so far On Wed, Apr 27, 2011 at 10:17 AM, Julien Krieger krieger.jul...@gmail.com wrote: Have you configured a range of directory number large enough? Do you have anything that could filter message exchanged between your phone and CUCM? In which state your phones are stucked? 2011/4/27 donny f f.faraday...@gmail.com yes i did , cause my UCM (HQ and BR-1 auto register ok0 On Wed, Apr 27, 2011 at 2:48 AM, Julien Krieger krieger.jul...@gmail.com wrote: Hi, Have you unchecked Auto-registration Disabled on this Cisco Unified Communications Manager under Cisco Unified CM on your CUCM? Have your configured the Starting Directory Number and Ending Directory Number under the same menu? That is probably it... Julien 2011/4/27 donny f f.faraday...@gmail.com hi exp, I have config following in BR-2 router, and enable auto -registration in UCM. However none of my BR-2 phone get register, and I have to do manual registration in UCM. wondering what i missed? Config - ip dhcp excluded-address 10.10.202.1 10.10.202.9 ip dhcp excluded-address 10.10.202.31 10.10.202.254 ip dhcp pool sc network 10.10.202.0 255.255.255.0 option 150 ip 10.10.210.10 default-router 10.10.202.1 interface Vlan400 ip address 10.10.202.1 255.255.255.0 ip helper-address 10.10.210.10 ip helper-address 10.10.210.11 h323-gateway voip bind srcaddr 10.10.202.1 end ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com http://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] how to configure cucme to support phoneview
Bruno, Please check IP Experts' website, I remember Vik had a blog about how to configure phoneview on it. Bo 2011/3/28 bruno bruno.juni...@gmail.com hello guys, great news Unified FX release their lab version. how to configure cucme to support phoneview? i can not find any tutorial on their website. could someone help？ Best Regards, bruno ___ 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] Cbarge in SRST nt working !
Hi, You might want to add ephone type under the ephone. Hope it helps, Bo On Mon, Mar 28, 2011 at 9:01 AM, Rahul Kapor rahul.kapo...@gmail.comwrote: Hi all , Cbarge in SRST not working here is my config ephone-dn-template 1 call-forward busy 914082026002 call-forward noan 914082026002 timeout 3 ephone-template 1 softkeys idle Redial Newcall Cfwdall ephone-dn 10 octo-line number conference ad-hoc ephone 1 privacy off device-security-mode none ephone 2 privacy off device-security-mode none telephony-service sdspfarm units 1 sdspfarm tag 1 HQ-CONF no privacy conference hardware srst mode auto-provision none srst ephone template 1 srst dn template 1 srst dn line-mode octo max-ephones 15 max-dn 15 ip source-address 14.160.116.40 port 2000 system message you are in fallback voicemail 914082026002 max-conferences 12 gain -6 moh music-on-hold.au multicast moh 239.1.1.1 port 16384 route 1.1.1.1 14.160.116.40 transfer-system full-consult create cnf-files version-stamp 7960 Mar 27 2011 01:04:02 Phones gets registered to SRST and shared line is seen on phone display. i created octo dn for conf and conf bridge is registered to SRST. Please let me know if i am missing any thing. thx, Rahul ___ 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] A 2nd Corporate Directory
Hi all, Is it possible to set up a Secondary corporate directory as an IP Phone service that coexist with the primary corporate directory in CUCM 7 or CUCM 8? Thank you, Bo ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Barge is not working in SRST
Try set your phone type under ephone. Bo On Sun, Jan 16, 2011 at 4:26 AM, Mritunjay Kumar mjs...@gmail.com wrote: Hi All , Cbarge in SRST is not working here is the config telephony-service sdspfarm units 2 sdspfarm tag 1 BR1-CNF no privacy conference hardware srst mode auto-provision none srst dn line-mode dual max-ephones 20 max-dn 20 ip source-address 14.160.116.40 port 2000 max-conferences 12 gain -6 transfer-system full-consult create cnf-files version-stamp Jan 01 2002 00:00:00 BR1# ephone-template 1 softkeys remote-in-use CBarge ephone-dn 1 octo-line number no-reg primary ephone-dn 2 octo-line number conference ad-hoc ephone-dn 3 dual-line number 3001 no-reg primary preference 5 ephone-dn 4 dual-line number 3002 no-reg primary preference 5 ephone 1 privacy off device-security-mode none mac-address 0026.CBBE.E8C9 ephone-template 1 button 1:3 2:1 ephone 2 privacy off device-security-mode none mac-address 0026.CBBE.EC4F ephone-template 1 button 1:4 2:1 hardware conf is registered . privacy is disabled under telephony service and in ephone. CME version 7.1 any missing config here ? Please suggest. Regards, Mritunjay ___ 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 Background Issues
In my case, the name of folder makes a difference. Some phones use 320x212x12, others use 320x212x16, and etc. Make sure your folder name and path in the list.xml set to what your phones are looking for. Regards, Bo On Sat, Aug 21, 2010 at 6:36 PM, Cyrus cyrus@gmail.com wrote: Hi, Every time worked for with the method: reload the router after you put tftp-server commnads for commands to kick in ,it's sort of like restarting TFTP service in CUCM! Cheers, On Sun, Aug 22, 2010 at 9:08 AM, Cisco CCIE ccieforl...@gmail.com wrote: I am using a 7961. This is what I see Aug 21 17:28:49.131: TFTP: Looking for Desktops/320x196x4/List.xml Ironic is that after about 5-10 min it worked! Does it take that amount of time? On Sat, Aug 21, 2010 at 6:02 PM, Daniel Berlinski dberlin...@gmail.comwrote: Hello What phone are you using? What happens with your debug tftp events? Is the phone looking for the path and files you have uploaded? Sometime you may need to adjust your tftp-server settings with the alias command cheers On Sun, Aug 22, 2010 at 8:01 AM, Cisco CCIE ccieforl...@gmail.comwrote: Yup that's the process I have always followed but end result is always hit and miss. Was wondering who else has ran into similar issues? On 8/21/10, Ashar Siddiqui siddas...@gmail.com wrote: Follow this process and you will never have a miss. At CME router do the following: Ping the tftp server and check connectivity Check if there is a directory on flash like Desktops/320x196x4/List.xml if not then make a directory Create directory in flash “mkdir flash:/Desktops/320x196x4” Copy files across copy tftp://10.10.210.5/List.xml flash:Desktops/320x196x4/List.xml copy tftp://10.10.210.5/small.png flash:Desktops/320x196x4/small.png copy tftp://10.10.210.5/small.png flash:Desktops/320x196x4/small.png Show the path to ephones tftp-server flash:Desktops/320x196x4/List.xml tftp-server flash:Desktops/320x196x4/large.png tftp-server flash:Desktops/320x196x4/small.png Reset ephones Go into Phone Settings à Preferences à Background image and select the new image Open debug tftp events and you will see following on router R3: May 23 15:44:32.367: TFTP: Looking for Desktops/320x196x4/List.xml May 23 15:44:32.367: TFTP: Opened flash:Desktops/320x196x4/List.xml, fd 8, size 152 for process 294 May 23 15:44:32.511: TFTP: Finished flash:Desktops/320x196x4/List.xml, time 00:00:00 for process 294 May 23 15:44:32.907: TFTP: Looking for Desktops/320x196x4/small.png May 23 15:44:32.911: TFTP: Opened flash:Desktops/320x196x4/small.png, fd 8, size 7196 for process 294 May 23 15:44:35.063: TFTP: Finished flash:Desktops/320x196x4/small.png, time 00:00:02 for process 294 May 23 15:44:39.083: TFTP: Looking for Desktops/320x196x4/large.png May 23 15:44:39.087: TFTP: Opened flash:Desktops/320x196x4/large.png, fd 8, size 73628 for process 294 May 23 15:45:00.323: TFTP: Finished flash:Desktops/320x196x4/large.png, time 00:00:21 for process 294 http://tinyurl.com/39cu8eq Ash Cisco CCIE wrote: OK so it appears that this has been happening with others as well. I did a search but none had it resolved. I had this working and then decided to redo the scenario but now it just won't work. Here are all the configurations just incase someone asks for it. Is there a bug with CME that makes this happen? i have never had ANY issues with background images in CUCM but CME is always a hit and a miss. Any help would be HIGHLY appreciated! tftp-server flash:Desktops/320x196x4/List.xml tftp-server flash:Desktops/320x196x4/phonelogoTN.png tftp-server flash:Desktops/320x196x4/phonelogo.png R3#dir Directory of flash:/Desktops/320x196x4/ 88 -rw- 158 Aug 21 2010 16:02:10 +00:00 List.xml 82 -rw- 14567 Aug 21 2010 15:57:08 +00:00 phonelogo.png 89 -rw-3293 Aug 21 2010 15:57:40 +00:00 phonelogoTN.png R3#more List.xml CiscoIPPhoneImageList ImageItem Image=TFTP:Desktops/320x196x4/phonelogoTN.png URL=TFTP:Desktops/320x196x4/phonelogo.png/ /CiscoIPPhoneImageList Aug 21 16:42:32.193: TFTP: Server request for port 49223, socket_id 0x4B6E1B6C for process 351 Aug 21 16:42:32.193: TFTP: read request from host 10.10.202.53(49223) via Vlan400 Aug 21 16:42:32.193: TFTP: Looking for Desktops/320x196x4/List.xml Also I do have the PHONE TYPE under ephones. Thanks in advance! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Sent from my mobile device ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding
Re: [OSL | CCIE_Voice] bad problem in configuring isdn pri
Do you need to set isdn incoming-voice voice on D channel? On Thu, Jul 22, 2010 at 2:49 AM, Akbar Ali ccie...@gmail.com wrote: Dear all , I am getting unexpected error while configuring isdn pri E1 , that i am not able to understand as provider says everything is right from their side what to do please help me. following is my configuration and errors also ... i tried self test on pri... controller E1 0/1/0 pri-group timeslots 1-31 description +BSNL PRI+ interface Serial0/1/0:15 no ip address encapsulation ppp dialer rotary-group 1 dialer-group 1 isdn switch-type primary-net5 no peer default ip address ppp authentication chap interface Dialer1 ip address 10.130.253.254 255.255.255.0 encapsulation ppp no ip mroute-cache dialer in-band dialer idle-timeout 9 dialer map ip 10.130.253.252 name FIS_HCBLROUTER broadcast dialer load-threshold 1 either dialer-group 1 HO-Rtr#sh log Syslog logging: enabled (0 messages dropped, 105 messages rate-limited, 0 flushes, 0 overruns, xml disabled, filtering disabled) No Active Message Discriminator. No Inactive Message Discriminator. Console logging: level debugging, 55360 messages logged, xml disabled, filtering disabled Monitor logging: level debugging, 0 messages logged, xml disabled, filtering disabled Buffer logging: level debugging, 55463 messages logged, xml disabled, filtering disabled Logging Exception size (4096 bytes) Count and timestamp logging messages: disabled Persistent logging: disabled Trap logging: level informational, 741 message lines logged Log Buffer (4096 bytes): *Jul 22 09:05:01.697: ISDN Se0/1/0:15 Q921: User RX - RRp sapi=0 tei=0 nr=16 *Jul 22 09:05:01.701: ISDN Se0/1/0:15 Q921: User TX - RRf sapi=0 tei=0 nr=16 *Jul 22 09:05:11.697: ISDN Se0/1/0:15 Q921: User RX - RRp sapi=0 tei=0 nr=16 *Jul 22 09:05:11.697: ISDN Se0/1/0:15 Q921: User TX - RRf sapi=0 tei=0 nr=16 *Jul 22 09:05:21.697: ISDN Se0/1/0:15 Q921: User TX - RRp sapi=0 tei=0 nr=16 *Jul 22 09:05:21.701: ISDN Se0/1/0:15 Q921: User RX - RRp sapi=0 tei=0 nr=16 *Jul 22 09:05:21.701: ISDN Se0/1/0:15 Q921: User TX - RRf sapi=0 tei=0 nr=16 *Jul 22 09:05:21.705: ISDN Se0/1/0:15 Q921: User RX - RRf sapi=0 tei=0 nr=16 *Jul 22 09:05:31.697: ISDN Se0/1/0:15 Q921: User RX - RRp sapi=0 tei=0 nr=16 *Jul 22 09:05:31.697: ISDN Se0/1/0:15 Q921: User TX - RRf sapi=0 tei=0 nr=16 *Jul 22 09:05:41.433: ISDN Se0/1/0:15 EVENT: UserIdle: callid 0x80B9 received IS DN_CALL (0x0) *Jul 22 09:05:41.433: ISDN Se0/1/0:15 EVENTd: UserIdle: Call to 2320635 at 64 Kb /s *Jul 22 09:05:41.433: ISDN Se0/1/0:15 EVENTd: isdn_get_guid: Cannot allocate a G UID (5) *Jul 22 09:05:41.433: ISDN Se0/1/0:15 EVENT: process_pri_call: call id 0x80B9, n umber 2320635, Guid 0026F91065D9, speed 64, call type DATA, redial No, CSM call No, pdata No *Jul 22 09:05:41.433: ISDN Se0/1/0:15 EVENTd: process_pri_call: No name in GTD *Jul 22 09:05:41.433: ISDN Se0/1/0:15 EVENTd: fill_cid_table_voice: Don't know c alling number for redial. *Jul 22 09:05:41.433: ISDN Se0/1/0:15 EVENTd: fill_cid_table_voice: Created entr y call_id 0x80B9, speed 64, remote 2320635, calling *Jul 22 09:05:41.433: ISDN Se0/1/0:15 EVENTd: Packet to CC Data *Jul 22 09:05:41.433: 4D000180B91604030800101804000300 *Jul 22 09:05:41.433: FF700900013233323036333504030800 *Jul 22 09:05:41.433: 101803000300 *Jul 22 09:05:41.433: ISDN Se0/1/0:15 EVENTd: calltrkr_setup_received: isdn_info =1732177424l, call_id=0x80B9 ORIGINATE *Jul 22 09:05:41.433: ISDN Se0/1/0:15 EVENTd: calltrkr_setup_received: calltrack er disabled *Jul 22 09:05:41.433: ISDN Se0/1/0:15 Q931: Sending SETUP callref = 0x0126 call ID = 0x80B9 switch = primary-net5 interface = User *Jul 22 09:05:41.433: ISDN Se0/1/0:15 Q921: User TX - INFO sapi=0 tei=0, ns=16 nr=16 *Jul 22 09:05:41.433: ISDN Se0/1/0:15 Q931: SETUP pd = 8 callref = 0x0126 Bearer Capability i = 0x8890 Standard = CCITT Transfer Capability = Unrestricted Digital Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA9839F Exclusive, Channel 31 Called Party Number i = 0x81, '2320635' Plan:ISDN, Type:Unknown *Jul 22 09:05:41.445: ISDN Se0/1/0:15 Q921: User RX - RR sapi=0 tei=0 nr=17 *Jul 22 09:05:41.453: ISDN Se0/1/0:15 Q921: User RX - INFO sapi=0 tei=0, ns=16 nr=17 *Jul 22 09:05:41.453: ISDN Se0/1/0:15 Q931: RELEASE_COMP pd = 8 callref = 0x812 6 Cause i = 0x829F - Normal, unspecified *Jul 22 09:05:41.453: ISDN Se0/1/0:15 Q921: User TX - RR sapi=0 tei=0 nr=17 *Jul 22 09:05:41.453: ISDN EVENTd: cc_clear_free_list freeing 0x673576E0 *Jul 22 09:05:41.453: ISDN Se0/1/0:15 EVENT: process_rxstate: ces/callid 1/0x80B 9 calltype 1 CALL_REJECTION *Jul
[OSL | CCIE_Voice] Frame-relay fragment question
HQ-BR1 bandwidth is 384K, I have the following config: map-class frame-relay AutoQoS-FR-Se0/0-201 frame-relay cir 384000 frame-relay bc 3840 frame-relay be 0 frame-relay mincir 384000 frame-relay fragment 480 service-policy output AutoQoS-Policy-Trust If I were to change the cir to 95% based on the QoS SNRD Then I would have: map-class frame-relay AutoQoS-FR-Se0/0-201 frame-relay cir 364800 frame-relay bc 3648 frame-relay be 0 frame-relay mincir 34800 frame-relay fragment 480 service-policy output AutoQoS-Policy-Trust Question: Should I also change the frame-realy fragment from 480 to 456? Why? Thank you! Bo ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Frame-relay fragment question
That's why I am confused. I have seen cases where some of them changed the value of fragment size, while the others did not. Thanks guys, I will keep looking for answers. Bo On Tue, Jun 29, 2010 at 7:23 AM, Graham Hopkins ghopk...@wolf-rock.co.ukwrote: I think not, the fragment size is related to the amount of data that can be placed on the wire in 10 ms which relates to line speed not CIR Graham Hopkins On 29 Jun 2010, at 15:17, Bo Gao bga...@gmail.com wrote: HQ-BR1 bandwidth is 384K, I have the following config: map-class frame-relay AutoQoS-FR-Se0/0-201 frame-relay cir 384000 frame-relay bc 3840 frame-relay be 0 frame-relay mincir 384000 frame-relay fragment 480 service-policy output AutoQoS-Policy-Trust If I were to change the cir to 95% based on the QoS SNRD Then I would have: map-class frame-relay AutoQoS-FR-Se0/0-201 frame-relay cir 364800 frame-relay bc 3648 frame-relay be 0 frame-relay mincir 34800 frame-relay fragment 480 service-policy output AutoQoS-Policy-Trust Question: Should I also change the frame-realy fragment from 480 to 456? Why? Thank you! Bo ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.comwww.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Frame-relay fragment question
Thank you Roger, I will leave it, then Bo 2010/6/29 Roger Källberg roger.kallb...@cygate.se Yes, that is correct. *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:* Berry, Matthew J. [mjbe...@krollontrack.com] *Skickat:* den 29 juni 2010 16:39 *Till:* Roger Källberg; Bo Gao; OSL *Ämne:* RE: [OSL | CCIE_Voice] Frame-relay fragment question I guess that makes sense. You’re not actually making the link slower, so the fragment size wouldn’t change. We’d only need to change the minCIR, CIR, and bc? *Matthew Berry*, *CCVP*, Sr. Unified Communications Engineer mjbe...@kroll.com david.ra...@kroll.com *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Roger Källberg *Sent:* Tuesday, June 29, 2010 9:33 AM *To:* Bo Gao; OSL *Subject:* Re: [OSL | CCIE_Voice] Frame-relay fragment question You shouldn't change the fragment size. Reason being that you want the fragment to be of a size that would give you a 10ms transmit delay in the event of congestion. 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:* Bo Gao [bga...@gmail.com] *Skickat:* den 29 juni 2010 16:17 *Till:* OSL *Ämne:* [OSL | CCIE_Voice] Frame-relay fragment question HQ-BR1 bandwidth is 384K, I have the following config: map-class frame-relay AutoQoS-FR-Se0/0-201 frame-relay cir 384000 frame-relay bc 3840 frame-relay be 0 frame-relay mincir 384000 frame-relay fragment 480 service-policy output AutoQoS-Policy-Trust If I were to change the cir to 95% based on the QoS SNRD Then I would have: map-class frame-relay AutoQoS-FR-Se0/0-201 frame-relay cir 364800 frame-relay bc 3648 frame-relay be 0 frame-relay mincir 34800 frame-relay fragment 480 service-policy output AutoQoS-Policy-Trust Question: Should I also change the frame-realy fragment from 480 to 456? Why? Thank you! Bo ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Strip + from calling number (SRST)
Hey Ash, What if I put the translation profile in SRST, rather than individual dial peer. I think it will work, will it? Bo On Sat, Jun 26, 2010 at 5:01 AM, Ashar Siddiqui siddas...@gmail.com wrote: You can Strip off '+' from a calling number when in SRST by using translation rules but any such rule will also affect your normal operation (not in SRST). I would advise if you want to take off '+' when in SRST, configure telephony service as srst mode auto prov all and then edit the ephone-dn. Ash Mark wrote: I'm trying to strip + from the calling number while a router is in SRST mode and phones dial outbound to the PSTN. I can't find a translation rule that does this. Does such a rule exist? Otherwise, how would you strip + from the e.164 format in SRST? ___ 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] Loading please wait - CUCM GUI error
I saw this once in a while in the CUCM 6, I suspected that it was b/c Tomcat lost connection with CUCM. Opening a new IE/Firefox window and re-login, then it would work out OK for me. Please try that and see if it works out for you. Bo On Fri, Jun 25, 2010 at 1:42 AM, Aman Chugh aman.ch...@gmail.com wrote: Which browser are you using. I have seen this before. It would not be a bad idea to clean up temp files of your browser like cookies and other browsing related data. HTH. Aman On Fri, Jun 25, 2010 at 1:14 PM, Ashar Siddiqui siddas...@gmail.comwrote: Hello all, I am having an issue while accessing GUI page of CUCM for one of the customer. CUCM version is 6.1.1.3101-1. When I enter username and password, the call manager accepts the credentials and then sits in “Loading please wait” state for indefinite time. Searched everywhere but couldn’t find any solution for it. Rebooted the box but no joy. All services are running. Restarted the Tomcat service as well. It’s just one call manager in the cluster. Any clue? Ash ___ 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] Transcoder/Conf MRG
If I want HQ, BR1, and BR2 all share one HD conference bridge and one HD transcoder, will it be better if I just leave these resources in the default null MRG, or assign them into the HQ_MRG, BR1_MRG, and BR2_MRG? Thanks, Bo ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Transcoder/Conf MRG
Hahahahah, Thank you Jeff. On Fri, Jun 25, 2010 at 6:23 PM, Jeff Price (jeffpric) jeffp...@cisco.comwrote: I would say better to put them in MRGs and then MRGLs. Although both would work, its better have control over who can access them. For example – HQ_R1_CONF_MRG, BR1_R2_XCODE_MRG Then create separate MRGLs with the same MRGs in them: HQ_MRGL – HQ_R1_CONF_MRG, BR1_R2_XCODE_MRG BR1_MRGL – HQ_R1_CONF_MRG, BR1_R2_XCODE_MRG However, for the exam purposes, it may just be easier to leave out for time J Jeff *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Bo Gao *Sent:* Friday, June 25, 2010 6:06 PM *To:* OSL *Subject:* [OSL | CCIE_Voice] Transcoder/Conf MRG If I want HQ, BR1, and BR2 all share one HD conference bridge and one HD transcoder, will it be better if I just leave these resources in the default null MRG, or assign them into the HQ_MRG, BR1_MRG, and BR2_MRG? Thanks, Bo ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Strip + from calling number (SRST)
If your ANI is +212555, and you wanted to keep 7 digits ANI, then it would be something like this: rule 1 /^\+212\(555\)/ /\1/ On Fri, Jun 25, 2010 at 10:03 PM, Mark m...@markholloway.com wrote: I'm trying to strip + from the calling number while a router is in SRST mode and phones dial outbound to the PSTN. I can't find a translation rule that does this. Does such a rule exist? Otherwise, how would you strip + from the e.164 format in SRST? ___ 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] + dialing in H323
Hi everyone, How do you configure voice translation rule in h323 gw so that it will translate the ANI into + format? Thank you, Bo ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] + dialing in H323
Aha! It is working, thank you very much! Bo On Mon, Jun 21, 2010 at 3:05 PM, Ashar Siddiqui siddas...@gmail.com wrote: Voice translation-rule 1 rule 1 // /+\0/ ! Voice translation-profile ANI-PLUS translate calling 1 Then under International dialpeer call this translation profile ~ translation-profile outgoing ANI-PLUS Ash Bo Gao wrote: Hi everyone, How do you configure voice translation rule in h323 gw so that it will translate the ANI into + format? Thank you, Bo -- ___ 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] Click to dial 7.0 not working
Did you enable click to call in the service parameters? Bo On Thu, Jun 17, 2010 at 2:38 AM, Azeem ahamed azeemo...@gmail.com wrote: hi All This question is not relevant to the CCIE voice Preparations but i looking out for all the help i could get. I am trying to use Click to Call with CCM 7.1.3 and whenever i give the User ID information it selects the EM profile. All the users are under EM. But when i try to dial it gives me error that Call Failed. Make sure the user is looged into Extension Mobility device. If problem presists please call CCM Administrator. anyone know whats wrong ? i have checked the following: - Webdialer service activated - User is associated with EM profile and logged in - In Click to Call application, went to Phone and verified a phone has been detected and associated ___ 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] MVA issue
Can you check Mobile Voice Access Number field in the Service Parameter to see if there are any prefix? Bo On Thu, May 27, 2010 at 10:21 AM, Leslie Meade lme...@signal.ca wrote: I know that general support is not the best option here. But I will ask.. I just noticed that my MVA is not working. Users can log into the system and attempt to dial, but the then get dead air Debugs show that some where I am appending an extra 7 to the remote destination profile, but I do not understand where. I am not using any transformation patterns, the gateway is not adding any digits.. The debug from vxml app on the gateway is showing correct numbers, debug ccapi is also showing correct, it is something on the Callmanager that is doing this. How can I track down what is adding the 7 ? 05/25/2010 20:09:57.870 CCM|SPROC :: stripAndPrependDigits- The number 777 is prepended with prefix 7, updated number=82284339|CLID::StandAloneClusterNID::CCM7-01LVL::DetailedMASK::ff 05/25/2010 20:09:57.870 CCM|SPROC getCtrlPid - callingNum=, inputCtrlPid=(1,100,175,1)|CLID::StandAloneClusterNID::x.x.x.xLVL::DetailedMASK::0800 05/25/2010 20:09:57.870 CCM|DbMobility: getMatchedRemDest starts: cnumber = |CLID::StandAloneClusterNID:: x.x.x.x LVL::DetailedMASK::ff 05/25/2010 20:09:57.870 CCM|DbMobility: getMatchedRemDest: full match case|CLID::StandAloneClusterNID:: x.x.x.x LVL::DetailedMASK::ff 05/25/2010 20:09:57.870 CCM|DbMobility: can't find remdest in map|CLID::StandAloneClusterNID::CCM7-01LVL::ErrorMASK::ff 05/25/2010 20:09:57.871 CCM|H225D::restart0_RSVPRegisterRes, CI=24083271, branch=0|CLID::StandAloneClusterNID:: x.x.x.x CT::1,100,152,1.1IP::10.1.1.5DEV::LVL::DetailedMASK::0800 ___ 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 Background Image Issue
I had the same problem a while back. Here is what I did to fix it: * TIP:* If you get the message: “ Selections unavailable” add these commands on the router: tftp-server flash:/Desktops/320x212x12/List.xml tftp-server flash:/Desktops/320x212x12/TN-logo.png tftp-server flash:/Desktops/320x212x12/logo.png after each of this you will get an error message like this: *Warning:* flash:Desktops/320x212x12/List.xml does not exist. Command retained. ***It seems that there is a bug on the CCME, but with these commands it is working!!!* * * * * Please try these and see if it works. Bo On Thu, May 27, 2010 at 4:46 PM, Salman Shaikh salman.shaik...@gmail.comwrote: Hi can any have any idea why my image is not showing. here is my config and debug ... CiscoIPPhoneImageList ImageItem Image=TFTP:Desktops/320x196x4/T-VOICE-7961.PNG URL=TFTP:Desktops/320x196x4/VOICE1-7961.PNG/ /CiscoIPPhoneImageList ! ! SC-R3#dir Directory of flash:/Desktops/320x196x4/ 53 -rw- 165 May 27 2010 22:33:34 +00:00 List.xml 54 -rw- 148026 May 27 2010 22:34:14 +00:00 VOICE1-7961.PNG 55 -rw- 10855 May 27 2010 22:34:36 +00:00 T-VOICE-7961.PNG 128034816 bytes total (44347392 bytes free) ! ! tftp-server flash:Desktops/320x196x4/T-VOICE-7961.PNG tftp-server flash:Desktops/320x196x4/VOICE1-7961.PNG tftp-server flash:Desktops/320x196x4/List.xml ! ! SC-R3(config)#do debug tftp events *May 27 22:49:38.068: TFTP: Looking for Desktops/320x196x4/List.xml SC-R3(config)# *May 27 22:49:42.068: TFTP: Looking for Desktops/320x196x4/List.xml SC-R3(config)# *May 27 22:49:46.064: TFTP: Looking for Desktops/320x196x4/List.xml SC-R3(config)# *May 27 22:49:50.064: TFTP: Looking for Desktops/320x196x4/List.xml SC-R3(config)# *May 27 22:49:54.068: TFTP: Looking for Desktops/320x196x4/List.xml SC-R3(config)# *May 27 22:49:58.064: TFTP: Looking for Desktops/320x196x4/List.xml SC-R3(config)# when i press settings User Preferences Background Image it shows me requesting selections but didn't see any image and then after a min try it shows selection Unavailable Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Gatekeeper Issue
Jeff, You were using trunks as terminal. I seem to remember that terminals can not specify a tech prefix. Change the type to VOIP-GW and give it a try. Bo On Tue, May 25, 2010 at 3:53 PM, Jeff Price (jeffpric) jeffp...@cisco.comwrote: Hi everyone, I am having trouble with my GK. I have made *Bold* what is the problem, but I can’t seem to understand why I’m having this issue. I configured a tech-prefix of 1# under the Trunk configuration page. Here is the config – gatekeeper zone local ZONE_1 asccie.com 10.5.200.1 zone prefix ZONE_1 1* gw-priority 10 CUCM_GK_TRUNK_2 zone prefix ZONE_1 1* gw-priority 9 CUCM_GK_TRUNK_1 zone prefix ZONE_1 1* gw-priority 0 BR2_R3_GW BR1_R2_GW zone prefix ZONE_1 44* gw-priority 10 BR2_R3_GW zone prefix ZONE_1 44* gw-priority 0 BR1_R2_GW CUCM_GK_TRUNK_2 CUCM_GK_TRUNK_1 gw-type-prefix 1#* default-technology no shutdown Here is the debug gatekeeper main 10 output: May 25 23:55:58.011: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup May 25 23:55:58.187: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup R1(config-gk)# May 25 23:56:00.115: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup May 25 23:56:00.115: ////GK/gk_rassrv_arq: arqp=0x4AE0FB04,crv=0x19, answerCall=0 May 25 23:56:00.115: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC May 25 23:56:00.115: //E69BDC8380B8/E69BDC8380BA/GK/gk_dns_query: No Name servers May 25 23:56:00.115: //E69BDC8380B8/E69BDC8380BA/GK/rassrv_get_addrinfo: (1#17752011001) Matched tech-prefix 1# May 25 23:56:00.115: //E69BDC8380B8/E69BDC8380BA/GK/rassrv_get_addrinfo: (1#17752011001) Matched zone prefix 1 and remainder 7752011001 May 25 23:56:00.115: ////GK/gk_rassrv_get_ingress_network: ARQ non-std ingress network = 1 May 25 23:56:00.115: //E69BDC8380B8/E69BDC8380BA/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x4AE06200 May 25 23:56:00.115: //E69BDC8380B8/E69BDC8380BA/GK/rassrv_arq_select_viazone: matched zone is ZONE_1, and z_invian R1(config-gk)#amelen=0 May 25 23:56:00.115: //E69BDC8380B8/E69BDC8380BA/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x4AE06200 May 25 23:56:00.115: //E69BDC8380B8/E69BDC8380BA/GK/rassrv_arq_select_viazone: matched zone is ZONE_1, and z_outvianamelen=0 May 25 23:56:00.115: ////GK/gk_rassrv_get_ingress_network: ARQ non-std ingress network = 1 May 25 23:56:00.115: //E69BDC8380B8/E69BDC8380BA/GK/rassrv_get_addrinfo: (1#17752011001) *tech-prefix gateway selection failed*. May 25 23:56:00.115: //E69BDC8380B8/E69BDC8380BA/GK/gk_rassrv_sep_arq: rassrv_get_addrinfo() failed (return code = 0x103) Here is the show gatekeeper call 10 output: May 26 00:02:15.899: ////GK/gk_call_new: src_endptp=0x4AE0F9F0, dst_endptp=0x0, src_pxp=0x0, dst_pxp=0x0, bw=160, crv=31, whichcrv=0x1, circuit=0x0, capacity=0x0, ret_callpp=0x4925F3F8 May 26 00:02:15.899: ////GK/gk_call_find_endpts: NOT_FOUND May 26 00:02:15.899: ////GK/gk_call_new: checking for default (CLI) carrier for sep endpt 0x4AE0F9F0 May 26 00:02:15.899: //C6CEF7C380D2/C6CEF7C380D4/GK/gk_call_delete: callp=4AB57F54 May 26 00:02:15.899: //C6CEF7C380D2/C6CEF7C380D4/GK/gk_call_delete: c_callstate 0x0, c_resbw1 0, resbw2 0, c_reszp1 0x0, c_reszp2 0x0 Here is the show gatekeeper endpoints output: R1(config-gk)#do show gatekeeper end GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags --- - --- - - - 10.5.201.1 1720 10.5.201.1 61751 ZONE_1VOIP-GW H323-ID: BR1_R2_GW Voice Capacity Max.= Avail.= Current.= 0 10.5.202.1 1720 10.5.202.1 52635 ZONE_1VOIP-GW H323-ID: BR2_R3_GW E164-ID: 3001 E164-ID: 3002 Voice Capacity Max.= Avail.= Current.= 0 172.21.51.204 37257 172.21.51.204 32858 ZONE_1TERM H323-ID: CUCM_GK_TRUNK_1 172.21.51.205 34279 172.21.51.205 32814 ZONE_1TERM H323-ID: CUCM_GK_TRUNK_2 Total number of active registrations = 4 (The reason why 3001 and 3002 are registering with GK is the fact that I am using the secondary command on CME. For some reason that is still letting 3001/3002 register with the GK). Thanks in advance for your help! [image: http://www.cisco.com/cisco/web/UK/images/emails/signaturetool/the_human_network_logo.jpg] *Jeff Price** Network Consulting Engineer - Unified Communications Practice* * * jeffp...@cisco.com Phone: *408-525-8293* Mobile: *408-204-4510*
Re: [OSL | CCIE_Voice] Single Cluster redundancy
SRST is your best bet. It is not possible to merge both East and West site into a single cluster b/c Cisco requires max. round trip delay between two CUCM servers with in a cluster to be less than 40ms. On Wed, May 19, 2010 at 12:49 PM, Mav nihil...@gmail.com wrote: Hello, I know that each cluster can have only one PUB ,my question is : in the scenario tha ACME company has two single Cluster one in NY and the other in LA ,both have one PUB and one SUB, the two site have 10 MB Point to Point, in the event that LA servers burn down, would it be possible have LA fail over to NY and have the phone register with the NY cluster ? or SRST is the only solution? Thanks. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Trancoding resources on CUBE
I think the number 90 was not b/c 45 (dspfarm) + 45 (sdspfarm), you had a value of 90 because each transcoder session consists of two transcoding streams between callers using transcode. Bo On Fri, May 7, 2010 at 3:33 AM, Matthew Berry ciscovoiceg...@gmail.comwrote: Earlier this week, I began Vol 2 Lab 1. In this lab, I configured transcoding resources on the CUBE. These resources were registered to the gateway itself, under telephony-service. I was messing around on a router this morning and found something confusing. If I define maximum sessions under the dspfarm profile as well as sdspfarm transcode sessions under telephony-service, the values seem to be considered independent of each other. I defined 45 maximum sessions on the dspfarm profile. Hower, when I run a show sdspfarm units, I get a total of 90 max-streams. The two commands appear to be summed up in this command. Can someone explain this to me? sccp local Loopback1 sccp ccm 192.168.99.1 identifier 1 version 7.0 sccp ! sccp ccm group 1 bind interface Loopback1 associate ccm 1 priority 1 associate profile 1 register RTR-XCODE signaling dscp ef ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 45 associate application SCCP ... telephony-service sdspfarm units 1 sdspfarm transcode sessions 45 sdspfarm tag 1 RTR-XCODE max-ephones 1 max-dn 1 ip source-address 192.168.99.1 port 2000 max-conferences 8 gain -6 transfer-system full-consult create cnf-files version-stamp Jan 01 2002 00:00:00 -- *Matthew Berry* *A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written* *Vitals:* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *Cert Stats:* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 ___ 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] Trancoding resources on CUBE
Matthew, MTP Dixieland is the hardware reference: MODEL_DIXIELAND_CFB = 52, // Cisco IOS Conference Bridge (HDV2) MODEL_DIXIELAND_SW_MTP = 83, // Cisco IOS Media Termination Point (HDV2) MODEL_DIXIELAND_MTP = 112, // Cisco IOS Media Termination Point (HDV2) On older equipment: MODEL_YOKO_CONF_BRIDGE = 51, // Conference Bridge WS-X6608 MODEL_YOKO_MTP = 111, // Media Termination Point Hardware I had done research on this in the past, I thought they were some kind of music :) Bo On Fri, May 7, 2010 at 5:51 AM, Berry, Matthew J. mjbe...@krollontrack.comwrote: Bo - You're right. I changed the max sessions value from 45 to 2. This is the new output from my show sdspfarm units: mtp-1 Device:RTR-XCODE TCP socket:[1] REGISTERED in SCCP ver 17/10 actual_stream:4 max_stream 4 IP:192.168.99.1 57105 MTP Dixieland keepalive 0 Supported codec: G711Ulaw G711Alaw G729 G729a G729ab max-mtps:1, max-streams:90, alloc-streams:4, act-streams:0 Two values to point out: MAX-STREAMS remains the same. This is the number of sdspfarm transcode sessions I specified under telephony-service. You're right, in that xocde/mtp resources are counted as multiples of two. Since I stated 45 sessions, it lists 90 streams. ALLOC-STREAMS reflects the max-sessions listed under the dspfarm profile section of my configuration. Since I entered 2 sessions, it displays 4 streams (again, multiples of two). What I need pay attention to is SESSIONS versus STREAMS. Lastly, what the heck is MTP Dixieland? (second line of the output). That's weird. Matthew Berry Digital Footprint: Twitter: ciscovoiceguru Skype: ciscovoiceguru 1st Lab Attempt: Aug 16th, 2010 From: ccie_voice-boun...@onlinestudylist.com [ ccie_voice-boun...@onlinestudylist.com] On Behalf Of Bo Gao [ bga...@gmail.com] Sent: Friday, May 07, 2010 7:22 AM To: Matthew Berry Cc: OSL Subject: Re: [OSL | CCIE_Voice] Trancoding resources on CUBE I think the number 90 was not b/c 45 (dspfarm) + 45 (sdspfarm), you had a value of 90 because each transcoder session consists of two transcoding streams between callers using transcode. Bo On Fri, May 7, 2010 at 3:33 AM, Matthew Berry ciscovoiceg...@gmail.com mailto:ciscovoiceg...@gmail.com wrote: Earlier this week, I began Vol 2 Lab 1. In this lab, I configured transcoding resources on the CUBE. These resources were registered to the gateway itself, under telephony-service. I was messing around on a router this morning and found something confusing. If I define maximum sessions under the dspfarm profile as well as sdspfarm transcode sessions under telephony-service, the values seem to be considered independent of each other. I defined 45 maximum sessions on the dspfarm profile. Hower, when I run a show sdspfarm units, I get a total of 90 max-streams. The two commands appear to be summed up in this command. Can someone explain this to me? sccp local Loopback1 sccp ccm 192.168.99.1 identifier 1 version 7.0 sccp ! sccp ccm group 1 bind interface Loopback1 associate ccm 1 priority 1 associate profile 1 register RTR-XCODE signaling dscp ef ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 45 associate application SCCP ... telephony-service sdspfarm units 1 sdspfarm transcode sessions 45 sdspfarm tag 1 RTR-XCODE max-ephones 1 max-dn 1 ip source-address 192.168.99.1 port 2000 max-conferences 8 gain -6 transfer-system full-consult create cnf-files version-stamp Jan 01 2002 00:00:00 -- Matthew Berry A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written Vitals: GVoice: +1.612.424.5044 Gmail: ciscovoiceg...@gmail.commailto:ciscovoiceg...@gmail.com Skype: ciscovoiceguru Twitter: ciscovoiceguru Cert Stats: Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] VLAN port speed and duplex
Hi guys, I am just starting my lab prep, and I am at lab 1A. When a port is combined with both voice and data, do we need to manually set port speed and duplex(i.e., 10/half) for the exam? Thank you, Bo ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
