Re: [OSL | CCIE_Voice] Gatekeeper VIAZONE - Could not find an IPIPGW
If the Voice service Voip commands are all configured...A restart has always helped me ... From: Boris boris.k...@gmail.com To: Steven forum.ccie.onlinestudyl...@nocer.net Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Wednesday, January 18, 2012 10:19 AM Subject: Re: [OSL | CCIE_Voice] Gatekeeper VIAZONE - Could not find an IPIPGW Hi Steve, Do you have This in your config? Voice service voip Allow h323 to h323 If not, add it and do no gateway/gateway Your cube should appear as H323 type in show gatek end. Sent from my mobile device, sorry for typos. --- Regards Boris On 18/01/2012, at 7:22, Steven forum.ccie.onlinestudyl...@nocer.net wrote: Hi there, i got some problems with my viazone (CUBE) at HQ-RTR. I already checked the Tech prefix match and it seems to succeed. But i'm clueless how to debug/resolve the Could not find an IPIPGW-problem. I also checked the dial-peers on HQ-RTR. Any help appreciated. Regards, Steven ! *** Begin tech details: HQ-RTR#debug gatekeeper main 10 ! tried to call from HQ (5002) to BR2 (3006) Jan 17 19:56:37.786: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Jan 17 19:56:37.786: ////GK/gk_rassrv_arq: arqp=0x48F0C08C,crv=0x7, answerCall=0 Jan 17 19:56:37.786: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/gk_dns_query: No Name servers Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_get_addrinfo: (3006) Tech-prefix match failed. Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_get_addrinfo: (3006) Matched zone prefix 3 and remainder 006 Jan 17 19:56:37.786: ////GK/gk_rassrv_get_ingress_network: returning default ingress network = 1 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x4793079C Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: matched zone is UCM, and z_invianamelen=0 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x47930A08 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: matched zone is UCME, and z_outvianamelen=4 Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone and z_outvianamep=CUBE Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_arq_select_viazone: Received ARQ for a zone (UCME) that has an outviazone (CUBE) specified. Pick an IP-IP gateway in that viazone. Jan 17 19:56:37.786: ////GK/gk_gw_select_ipipgw_random: zonep: 0x47930C74, tpp: 0x0, current_endpt: 0 Jan 17 19:56:37.786: ////GK/gk_gw_select_ipipgw_random: Selecting any IPIPGW. qelemp.head=0x46F0FE88, use_count=1, current_endpt=0 Jan 17 19:56:37.786: ////GK/gk_gw_select_ipipgw_random: qelemp=0x46F0FE88, loop_count=0 Jan 17 19:56:37.786: ////GK/gk_gw_select_ipipgw_random: Examining tgwp 0x46F253E0, g_supp_prots: 0x50 qelemp: 0x46F0FE88, loop_count: 1 Jan 17 19:56:37.786: ////GK/gk_gw_select_ipipgw_random: Searched through the entire gateway list. loop_count=0 Jan 17 19:56:37.786: ////GK/gk_gw_select_ipipgw_random: Could not find an IPIPGW. Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/rassrv_get_addrinfo(3006): Viazone gateway selection failed for zone CUBE Jan 17 19:56:37.786: //80AC69450700/80AC69450700/GK/gk_rassrv_sep_arq: rassrv_get_addrinfo() failed (return code = 0x805) HQ-RTR#show gatekeeper gw-type-prefix GATEWAY TYPE PREFIX TABLE = Prefix: 1#* (Default gateway-technology) Zone CUBE master gateway list: 10.10.110.1:1720 HQ-RTR Zone UCM master gateway list: 10.10.210.10:44248 gk-trunk_1 10.10.210.11:36641 gk-trunk_2 Prefix: 3#* Zone UCME master gateway list: 10.10.110.3:1720 BR2-RTR HQ-RTR#show gatekeeper zone prefix ZONE PREFIX TABLE = GK-NAME E164-PREFIX --- --- UCME 3... UCM 5... HQ-RTR#show running-config interface loopback 0 interface Loopback0 ip address 10.10.110.1 255.255.255.255 h323-gateway voip interface h323-gateway voip id CUBE ipaddr 10.10.110.1 1719 h323-gateway voip h323-id HQ-RTR h323-gateway voip tech-prefix 1# HQ-RTR#show running-config | section gatekeeper gatekeeper zone local UCM ipexpert.com zone local UCME ipexpert.com outvia CUBE zone local CUBE ipexpert.com zone prefix UCME 3... zone prefix UCM 5... gw-type-prefix 1#* default-technology no shutdown ___ For
[OSL | CCIE_Voice] VoiceView Under SRST
Hello Guys I have never any document related to this so seeking help from anyone who can guide me on this ..Whether if its even possible or not.. I have Branch 2 site phones registered with CUCM.. The Unity express module with Jtapi integration . I have configured voice view and it works fine.. Now when the phones go under SRST. Phones work fine.. CUE starts using SIP integration as fall back.. I have configured URL services undere tele-phony-service.. When someone calls Branch2 phone.. after 12 seconds the call goes to Unity express as expected..MWI works fine.. When i press the services button on the phone.. I can see my inbox.. but when i press listen ..i get error message .. I have tried every possible thing but nothing worked for me .. Can anyone share their experience with me to get Voiceview going under SRST.. Thanks a lot.. Regards ___ 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] CUE Voice-View
Hello Guys I have tried to soo many times but it has never worked perfectly for me .. Could anyone please help me with this .. here is my config...My cue is working fine.. I call press the message button and listen to messages .. Voice view is giving me grief.. can anyone please point my mistake here .. Thanks .. let me know if u can or want.. i can organize remote access session.. Thanks telephony-service privacy-on-hold conference hardware no auto-reg-ephone authentication credential phone cisco max-ephones 4 max-dn 20 no-reg both ip source-address 10.10.202.1 port 2000 url services http://10.10.202.2/voiceview/common/login.do url authentication http://10.10.202.1/CCMIP/authenticate.asp cisco cisco live-record 3650 voicemail 3600 mwi relay max-conferences 8 gain -6 web admin system name cme password cisco dn-webedit time-webedit transfer-system full-consult transfer-pattern .T ___ 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] CME Shared line and Hunstop Channel
Hello Guys I am working on this lab and for some reason its not working as expected ... The expected behavior is 5 Maximum calls to Number 3004. Phone 1 should be able to Handle 3 incoming calls .. 4th call should ring on Phone 2.. Phone 2 should be able to handle max 2 calls.. rest 3 should ring on Phone 1.. when i call 3004 from 3 different phones.. and answer all the 3 calls on phone 1..The fourth incoming call gives me busy signal straightaway..Iit doesnt ring on Phone 2..Which it should... What am i missing here ? Thanks in advance for your help.. Here is the config.. ephone-template 1 softkeys remote-in-use CBarge Newcall ephone 1 privacy off privacy-button device-security-mode none description +3432143001 mac-address 0026.0B5D.EDFA ephone-template 1 max-calls-per-button 5 busy-trigger-per-button 3 button 1:1 2:4 ! ! ! ephone 2 privacy off device-security-mode none description +3432143002 mac-address 0021.A02B.E1C5 ephone-template 1 max-calls-per-button 3 busy-trigger-per-button 2 button 1:2 2:4 3w1 ephone-dn 1 octo-line number 3001 no-reg primary description +3432143001 name SC Ph 1 ! ! ephone-dn 2 octo-line number 3002 no-reg primary description +3432143002 name SC Ph 2 ephone-dn 4 octo-line number 3004 description +3432143004 name Shared huntstop channel 5 ___ 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] BACD Drop Through MoH issue
Hello Everyone.. I think i had a very bad last night ...Finally i had a look at my own configs again.. I managed to fix this...Now when i dial 32143000.. the call goes to hunt group as expected.. If both users are busy, i can hear MOH as expected.. originally i put this command which was wrong..paramspace english location flash: After the change its all good.. My config looks like this now and its working.. service aa flash://bacdprompts/app-b-acd-aa-2.1.2.2.tcl paramspace english index 0 param number-of-hunt-grps 1 param drop-through-option 1 param handoff-string aa paramspace english language en param max-time-vm-retry 2 param aa-pilot 32143000 paramspace english location flash://bacdprompts/ param call-retry-timer 15 param max-time-call-retry 700 param voice-mail 3600 param service-name callq Thanks for your support.. Regards Amit From: amit batra batraji...@yahoo.com To: ccie voice ccie_voice@onlinestudylist.com Sent: Monday, September 12, 2011 12:21 PM Subject: [OSL | CCIE_Voice] BACD Drop Through MoH issue Hello Guys I dont know if this issue is related to my previous sent email or different..Anyways ... The question in the lab says Configure BR2 so that when dial 032143000 it should ring 3001 and 3002 . If both users are busy queue the call and MoH should be played...I configure BaCD drop through mode..When i call 032143000 BACD works as expected, both phone 3001 and then 3002 ring..If no one answers the call goes to Queue..BACD application keeps trying to ring these phones again and again.. The only thing which is not working is when the call goes in the Queue.. The PSTN user doesn't hear MoH.. If i call 32143001 from PSTN and press hold key.. PSTN user can hear MoH... Here is the configuration of my BACD...Can anyone help me spot the missing bit for Moh to work when the call is in the Queue.. Thanks in advance .. application service callq flash://bacdprompts/app-b-acd-2.1.2.2.tcl param queue-len 10 param aa-hunt1 3100 param number-of-hunt-grps 1 param queue-manager-debugs 1 ! service aa flash://bacdprompts/app-b-acd-aa-2.1.2.2.tcl paramspace english index 0 param number-of-hunt-grps 1 param drop-through-option 1 param handoff-string aa paramspace english language en param max-time-vm-retry 2 param aa-pilot 32143000 paramspace english location flash: param call-retry-timer 15 param max-time-call-retry 700 param voice-mail 3600 param service-name callq ! dial-peer voice 3000 pots service aa incoming called-number 32143000 direct-inward-dial port 0/0/0:15 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] MOh Issue
Hello Guys I am stuck with this strange thing and for some reason cant get Moh working on my CME BR2 router..It has always worked in the past Multicast and Unicast.. In this particular question, there are no Multicast requirement so i just configure moh music-on-hold.au command. When i call from PSTN phone to 32143001 and user 3001 put the call on hold i can hear MoH fine.. But when user 3001 calls 3002 (Internal call) .. All i hear is a beep...I have checked. the codec is G711u for sure. Can anyone please advice on what i am missing.. Thanks .. Here is the config .. telephony-service sdspfarm units 3 sdspfarm transcode sessions 4 sdspfarm tag 1 BR2-XCODER no auto-reg-ephone max-ephones 5 max-dn 20 no-reg ip source-address 10.10.202.1 port 2000 time-zone 23 voicemail 3600 max-conferences 8 gain -6 moh music-on-hold.au transfer-system full-consult create cnf-files version-stamp 7960 Sep 11 2011 11:04:38 ! ! ephone-dn 1 octo-line number 3001 no-reg primary description 032143001 name BR2 Phone 1 call-forward busy 3600 call-forward noan 3600 timeout 12 mwi sip ! ! ephone-dn 2 octo-line number 3002 no-reg primary description 032143002 name BR2 Phone 2 call-forward busy 3600 call-forward noan 3600 timeout 12 mwi sip ! ! ephone-dn 3 octo-line number 3003 no-reg primary description 032143003 name BR2 Phone 3 ! ! ephone 1 no multicast-moh device-security-mode none description 032143001 mac-address 0026.0B5D.EDFA button 1:1 ! ! ! ephone 2 no multicast-moh device-security-mode none description 032143002 mac-address 0021.A02B.E1C5 blf-speed-dial 1 5002 label 5002 button 1:2 2w1___ 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] BACD Drop Through MoH issue
Hello Guys I dont know if this issue is related to my previous sent email or different..Anyways ... The question in the lab says Configure BR2 so that when dial 032143000 it should ring 3001 and 3002 . If both users are busy queue the call and MoH should be played...I configure BaCD drop through mode..When i call 032143000 BACD works as expected, both phone 3001 and then 3002 ring..If no one answers the call goes to Queue..BACD application keeps trying to ring these phones again and again.. The only thing which is not working is when the call goes in the Queue.. The PSTN user doesn't hear MoH.. If i call 32143001 from PSTN and press hold key.. PSTN user can hear MoH... Here is the configuration of my BACD...Can anyone help me spot the missing bit for Moh to work when the call is in the Queue.. Thanks in advance .. application service callq flash://bacdprompts/app-b-acd-2.1.2.2.tcl param queue-len 10 param aa-hunt1 3100 param number-of-hunt-grps 1 param queue-manager-debugs 1 ! service aa flash://bacdprompts/app-b-acd-aa-2.1.2.2.tcl paramspace english index 0 param number-of-hunt-grps 1 param drop-through-option 1 param handoff-string aa paramspace english language en param max-time-vm-retry 2 param aa-pilot 32143000 paramspace english location flash: param call-retry-timer 15 param max-time-call-retry 700 param voice-mail 3600 param service-name callq ! dial-peer voice 3000 pots service aa incoming called-number 32143000 direct-inward-dial port 0/0/0:15 ___ 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] Cisco TAPS
Hello guys recentlly i have heard some thing about TAPS.. Well i myslef have never worked with TAPS so got scared a little bit . On Cisco website i was looking for documentation but couldnt find a step by step guide ..I have found this document which i kinda what i was looking. So thought of sharing with everyone here .. Some people might find this link useful.. http://wiki.blindhog.net/index.php/Sandbox:How_to_use_TAPS#Install_TAPS_Plugin_on_UCCX_Server I am going to try this when i get home.. regards Amit ___ 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] VPIM Vol2 Lab 2 Question 8.3
Hello Everyone I have configured VPIM as mentioned in the lab and using the same steps mentioned in the video by Vik..But i am unable to send message from any of the locations. When i try to send message from the phone, i get the confirmation that message sent but no MWI no message in the inbox..I also ran the trace on CUE and not even a single line output...Can anyone please let me know what is missing from the config .. I am using my own hardware my own hardware so the naming convention is not 100% same as per the work book..But here are the details. CUE IP = 10.10.202.2 Hostname = cue Domain name = cue.com UC IP = 10.10.210.13 Hostname = CUC Domain name = proctorlabs.com The DNS is configured with Host record and MX records and resolution is working fine from CUE and UC both.. I have tested end to end ping by using hostnames . The location on CUE are configured as below network location id 212 email domain proctorlabs.com name cuc end location network location id 331 email domain cue.com name cue end location network local location id 331 UC configuration.. Edit Connection Location Display Name* Host Address SMTP Domain Name Connection Version VPIM location Edit VPIM Location Display name* Dtmf Access ID* Partition Search Scope Remote VPIM Domain Name* IP Address* Voice Name Prefixes Remote phone prefix Cisco Connection phone prefix Automatic Gain Control (AGC) Settings Enable AGC Audio format conversion Incoming messages Outbound messages Message Settings Sender's recorded name Enable Outgoing Secure Messages Enable Outgoing Private Messages Allow Blind Addressing Thanks in advance ..___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Call Preservation Issue
try this http://ccieash.wordpress.com/2010/06/25/h323-call-preservation/ might help you... --- On Wed, 6/1/11, CCIE Voice ccievoicelab.c...@gmail.com wrote: From: CCIE Voice ccievoicelab.c...@gmail.com Subject: Re: [OSL | CCIE_Voice] Call Preservation Issue To: amit batra batraji...@yahoo.com Cc: ccie_voice@onlinestudylist.com Date: Wednesday, June 1, 2011, 4:34 AM Sorry, should have been more clear. Using H.323. The call preservation feature of H323 applies when the gateway loses connectivity to Call Manager only, so that the RTP stream is preserved to the phone, but here the RTP stream itself is being interrupted. On May 31, 2011, at 6:58 PM, amit batra batraji...@yahoo.com wrote: What type of protocol you are using ? H323 or MGCP ? With MGCP i dont think if there is any way .. H323 can preserve call .. --- On Wed, 6/1/11, CCIE VOICE ccievoicelab.c...@gmail.com wrote: From: CCIE VOICE ccievoicelab.c...@gmail.com Subject: [OSL | CCIE_Voice] Call Preservation Issue To: ccie_voice@onlinestudylist.com Date: Wednesday, June 1, 2011, 3:03 AM Hello Experts An interesting problem to solve for which I'm having a hard time figuring out the solution myself. We have a Call Manager node and IP Phones in one location and ingress voice gateways in other location. Both locations are connected over redundant WAN links. When one WAN link fails, it takes upto 5 seconds for the redundant WAN link to kick in. Issue is, if there is an active call between the gateway and the IP Phone, the call simply drops when a WAN issue occurs. Is there a way to preserve the RTP stream on the GW itself that can be resumed once the redundant WAN link takes over? Any input is appreciated! -Inline Attachment Follows- ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Vpim Testing in Home lab
Hello Everyone i know a lot of people study for CCIE Voice using their own hardware ( I am doing the same). I was just wondering that how people practice VPIM ? I use same version of CUCM and unity as shown in Vik's video's.. When ever i try to configure VPIM i get license violation error message .. Is there any workaround ? or i will need to buy license for that ?? Any advice on that ? Thanks in advance. Regards Amit ___ 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] Vol2 Lab 2 Supplementary services with GK Trunk
Hello Everyone. I have finished IPexperts Volume 2 lab 2 .. I managed to finish everything apart from VPIM and Supplementary services ...VPIM is a license issue so not worried .. I have configured a GK trunk between CUCM and HQ router (Gatekeeper) and BR2 router. all endpoints are registered to the gatekeepers. No probs till here .. Call from CUCM to CME and vice versa are working. The problem i am facing is , when i make a call from CUCM phone 5001 (SCCP) to a CME phone 3002 (SCCP) audio works fine. i can press hold button on the CUCM phone. When i do that on CME phone i hear beep. but when i press resume on CUCM phone, CME phone keep's giving that beep sound. when i press hold button on my CME and resume , audio start to flow again.. I have configured software MTP on HQ router. Device pool assigned to the GK-Trunk and this software MTP is the same . On GK-Trunk MRGL is assigned .. Media Termination Point Required (ON) Retry Video Call as Audio (ON) Wait for Far End H.245 Terminal Capability Set (ON) Inbound faststart enabled (ON) when i make a call from any device , i can see that my IOS MTP is invoked and participating in the call .. show sccp connections Am i missing anything here ? or do i need to enable anything else..? I hope i am making some sense.. If the question is not clear please let me know. and 1.30 am i cannot write anything more than this.. Thanks in advance .. Regards Amit ___ 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] Voiec View Authentication issue
Hi Shingei This morning I tried this and it works after the change. Thanks a lot for your help. Regards Amit Sent from my iPhone On 22/03/2011, at 5:43 PM, ShinGei Yong shingei.y...@gmail.com wrote: Can you change your authentication URL to the below: url authentication http://cme-ip-address/CCMCIP/authenticate.asp Thanks Shingei On Tue, Mar 22, 2011 at 5:22 AM, Jimmy batraji...@yahoo.com wrote: Hi I have tried that as well. After doing that create cnf. Even restarted the routers and power off on phones as well. Still no luck. Regards Sent from my iPad On Mar 22, 2011, at 3:04 AM, ShinGei Yong shingei.y...@gmail.com wrote: Can you change your authentication path? Instead of pointing to your 202.128, changed it to your UCME ip address, which is your 202.1 Thanks Shingei On Mon, Mar 21, 2011 at 9:18 PM, Prashant Patel prashantpatel...@gmail.com wrote: Hi Amit, Do you have the following in the CME config no ip http secure-server ip http server ip http path flash: (or /gui - path where u have the cme files) ip http auth local HTH Prashant On Mon, Mar 21, 2011 at 7:19 AM, amit batra batraji...@yahoo.com wrote: Hello Everyone .. I have searched everywhere and tried all commands but this thing isnt working..CUE is integrated with CME (BR2). everything is working as expected.. I have configured Voiceview . When i press the services button. I can login. i can see messages in the inbox. but when i press to listen , i get this error message .. Authentication error. Report this error to your system administrator I have all the required license on the CUE .. show software licenses Installed license files: - voicemail_lic.sig : 12 MAILBOX LICENSE Core: - Application mode: CCME - Total usable system ports: 8 Voicemail/Auto Attendant: - Max system mailbox capacity time: 6000 - Default # of general delivery mailboxes: 5 - Default # of personal mailboxes: 12 - Max # of configurable mailboxes: 17 Interactive Voice Response: - Max # of IVR sessions: Not Available Languages: - Max installed languages: 5 - Max enabled languages: 5 Here is the CME configuration telephony-service sdspfarm units 1 sdspfarm transcode sessions 2 sdspfarm tag 1 BR2TRANS no auto-reg-ephone authentication credential admin 1234 load 7960-7940 P00308000500 max-ephones 3 max-dn 10 ip source-address 10.10.202.1 port 2000 timeouts interdigit 3 system message BR2 CME url services http://10.10.202.128/voiceview/common/login.do url authentication http://10.10.202.128/voiceview/authentication/authenticate.do date-format dd-mm-yy live-record 3609 voicemail 3600 max-conferences 8 gain -6 moh music-on-hold.au web admin system name admin password 1234 dn-webedit time-webedit transfer-system full-consult secondary-dialtone 0 create cnf-files version-stamp 7960 Mar 22 2011 00:56:56 Any ideas ??? Thanks in advance .. ___ 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] Voiec View Authentication issue
Hello Everyone .. I have searched everywhere and tried all commands but this thing isnt working..CUE is integrated with CME (BR2). everything is working as expected.. I have configured Voiceview . When i press the services button. I can login. i can see messages in the inbox. but when i press to listen , i get this error message .. Authentication error. Report this error to your system administrator I have all the required license on the CUE .. show software licenses Installed license files: - voicemail_lic.sig : 12 MAILBOX LICENSE Core: - Application mode: CCME - Total usable system ports: 8 Voicemail/Auto Attendant: - Max system mailbox capacity time: 6000 - Default # of general delivery mailboxes: 5 - Default # of personal mailboxes: 12 - Max # of configurable mailboxes: 17 Interactive Voice Response: - Max # of IVR sessions: Not Available Languages: - Max installed languages: 5 - Max enabled languages: 5 Here is the CME configuration telephony-service sdspfarm units 1 sdspfarm transcode sessions 2 sdspfarm tag 1 BR2TRANS no auto-reg-ephone authentication credential admin 1234 load 7960-7940 P00308000500 max-ephones 3 max-dn 10 ip source-address 10.10.202.1 port 2000 timeouts interdigit 3 system message BR2 CME url services http://10.10.202.128/voiceview/common/login.do url authentication http://10.10.202.128/voiceview/authentication/authenticate.do date-format dd-mm-yy live-record 3609 voicemail 3600 max-conferences 8 gain -6 moh music-on-hold.au web admin system name admin password 1234 dn-webedit time-webedit transfer-system full-consult secondary-dialtone 0 create cnf-files version-stamp 7960 Mar 22 2011 00:56:56 Any ideas ??? Thanks in advance .. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Calling and Called Party Number Type
Watch Vik Malhi's video .. He has explaines this very well.. Answer to your question is in his video.. --- On Fri, 3/4/11, Ccie Voice v.c...@yahoo.com wrote: From: Ccie Voice v.c...@yahoo.com Subject: Re: [OSL | CCIE_Voice] Calling and Called Party Number Type To: Matthew Berry matt...@ciscovoiceguru.com, ccie_voice@onlinestudylist.com Date: Friday, March 4, 2011, 1:09 PM Thank you all, I know that debug will help me to make sure that I am sending the configured values. but not this is not my concern. I need to know how to set these values? is there any rule I should follow or not? From: Matthew Berry matt...@ciscovoiceguru.com To: ccie_voice@onlinestudylist.com Sent: Fri, March 4, 2011 4:38:15 AM Subject: Re: [OSL | CCIE_Voice] Calling and Called Party Number Type For sure, Adam! debug isdn q931 was my best friend in the lab. A close runner-up is debug voip dialpeer to make sure the correct dial-peer is being selected. Thanks! Matthew Berry, CCIE #26721 (Voice) Email: thematthewbe...@gmail.com Twitter: http://twitter.com/CiscoVoiceGuru Tech Blog: http://ciscovoiceguru.com On 3/3/11 6:23 PM, adam compton wrote: If you run debug isdn q931 on the gateway the call is going out, you can confirm what Call Manager is sending to the PSTN. On Thu, Mar 3, 2011 at 1:52 PM, Ccie Voice v.c...@yahoo.com wrote: Thank you all for your reply, I just need to know if the PSTN router in the LAB will accept the call or no if it is not set to the proper value. If the PSTN router will not accept the call then it is OK I can play with these values and solve the problem. But the problem if the PSTN router accepts all calls based on called party number and later on the proctor will check if you set the values correctly or not. for me what I understood before is the way that Roger sent. (thank you Roger) Regards, From: Roger Källberg roger.kallb...@cygate.se To: Ccie Voice v.c...@yahoo.com; CCIE Study ccie_voice@onlinestudylist.com Sent: Thu, March 3, 2011 6:41:12 PM Subject: SV: [OSL | CCIE_Voice] Calling and Called Party Number Type Hi, You need to look at this from the originating endpoint and the outgoing gateway. For a more detailed explanation see my response in line with your mail. Sincerely Roger Källberg CCIE #26199 (Voice) Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ Från: Ccie Voice [v.c...@yahoo.com] Skickat: den 3 mars 2011 02:49 Till: CCIE Study Ämne: [OSL | CCIE_Voice] Calling and Called Party Number Type Hi All, I am a little bit confused about how to set the value for Calling and Called Party Number Plan. let us say HQ Phone 1 Calls local Call in this case I think I have to set: Calling Party Number Type to: Subscriber. Called Party Number Type to: Subscriber. This is correct What about Long Distance: Calling Party Number Type to: Subscriber or National From the perspective of caller and VGW this is a call that came from a local site , aka it's subscriber Called Party Number Type to: National From the perspective of called and VGW this is a call goes to a remote phone, aka it's national it will be more complicated if we need to use TEHO, So if HQ Phone 1 calls BR1 Local PSTN number what I should set the values? Long Distance, using BR1 Router Calling Party Number Type to: Subscriber or National From the perspective of caller and VGW this is a call that came from a remote site , aka it's national Called Party Number Type to: National or Subscriber From the perspective of called and VGW this is a call goes to a local phone, aka it's subscriber Long Distance, backup for BR1 using HQ Router Calling Party Number Type to: Subscriber or National From the perspective of caller and VGW this is a call that came from a local site , aka it's subscriber Called Party Number Type to: National or Subscriber I am using BR1 Router From the perspective of called and VGW this is a call goes to a remote phone, aka it's national 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 -Inline Attachment Follows- ___ 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] Does anybody read my messages?
Hello guys. could you please do everyone a favour of sending unicast reply. Instead of sending a broadcast reply all. Thanks. Regards Amit Sent from my iPhone On 25/02/2011, at 9:55 PM, Mohamed Gazzaz mgaz...@hotmail.com wrote: Yes Date: Thu, 24 Feb 2011 23:52:32 -0600 From: danie...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Does anybody read my messages? Hello all, I'm very sorry if I'm disturbing you but for some reason I feel nobody receives my posts. Does anyone read this? Thanks a lot! Daniel G. ___ 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] T1 Pri Issue
hello guys correct me if i am wrong .. under the t1 controller pri-group command with service mgcp prefix is missing apart from that .. Have you enabled MGCP on the router ? --- On Fri, 1/7/11, givemeccievoice2...@gmail.com givemeccievoice2...@gmail.com wrote: From: givemeccievoice2...@gmail.com givemeccievoice2...@gmail.com Subject: Re: [OSL | CCIE_Voice] T1 Pri Issue To: 'Deepak sidana' sidana_dee...@yahoo.com, ccie_voice@onlinestudylist.com Date: Friday, January 7, 2011, 3:26 AM Do you have this GW configured on CUCM? Is the gateway showing registered on CUCM? Do you have mgcp configured/enabled on the router? Have you bounced the MGCP (no mgcp/mgcp) after configuring the pri? You will see TEI Assigned until you have successfully configured all aspects of the MGCP GW. Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Deepak sidana Sent: Thursday, January 06, 2011 1:25 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] T1 Pri Issue Hi All, I am trying to connect the T1 from Br1-RTR to PSTN-WAN. Only when i use service mgcp, under controller, layer 2 isdn staus as TEI_ASSIGNED. At PSTN-WAN Router, i am using isdn protocol-emulate network under s0/0/0:23 Branch1 Config:- BR1-RTR#sh isdn sta Global ISDN Switchtype = primary-net5 ISDN Serial0/0/0:23 interface dsl 0, interface ISDN Switchtype = primary-net5 Layer 1 Status: ACTIVE Layer 2 Status: TEI = 0, Ces = 1, SAPI = 0, State = TEI_ASSIGNED ! controller T1 0/0/1 framing esf linecode b8zs cablelength long 0db ! interface Serial0/0/0:23 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn bind-l3 ccm-manager isdn incoming-voice voice no cdp enable Please share you experince, if some one faced the same issue. ThanksRgds Deepak Sidana. -Inline Attachment Follows- ___ 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] IP Blue Phone with Windows 7
Hello Everyone.. I just wanna check one with with people in this forum..Does IP Blue soft phone works fine wwith Windows 7 operating system or its just me having troubles.. I have tried installing uninstalling and everything.. It doesnt work some time and the error i get is Set the primary call manager IP Out of 10 it works 1 time without me doing anything... Is this a known problem ?? Has anyone exprienced this before ? Thanks Have a great evening.. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCCX 7.0(1) Error - Cannot connect to the IPPhone Agent service
Dear Tam, Today i have tried this lab..let me explain you first on what i have done so far and whats working and whats not working.. I created a CSQ, resources, agents, ..Everything works...Call from PSTN to 2123945400 get answered by UCCX..I have a queue name tech... the agent login on the computer with a CIPC (1st lab) 7960 (2nd lab)These both scenarios work fine.. Now i create a phone service CUCM and assign it to the phone.. within service URL i define the extension of IPCC agent username and password.. When i hit the button i get IPPA.. i select and then i get the message The password is invalid.. The Location you have mention in the email below, i have captured logs from the log file...The part of it is mentioned below... I dont know why it doesnt work on IP phone.. It works with CAD... same username and password.. VERSION INFO: Manifest: VoiceBrowser.jar: Product Version Build 7.0(1.168). VERSION INFO: Manifest: js.jar: Product Version Build . VERSION INFO: Manifest: idl.jar: Product Version Build . VERSION INFO: Manifest: CiscoDocumentSteps.jar: Product Version Build 7.0(1.168). VERSION INFO: Manifest: CiscoSessionSteps.jar: Product Version Build 7.0(1.168). VERSION INFO: Manifest: StepsIVR.jar: Product Version Build 7.0(1.504). VERSION INFO: Manifest: CiscoUserSteps.jar: Product Version Build 7.0(1.168). VERSION INFO: Manifest: CiscoGrammarSteps.jar: Product Version Build 7.0(1.168). VERSION INFO: Manifest: SplkStd4J.jar: Product Calabrio Internal Shared Library Version 1.0 Build 2.0.13. VERSION INFO: Manifest: log4j.jar(org/apache/log4j/): Product Version Build 1.2.14. 2010-09-22 18:23:03,116 INFO IPPA0001 IPPA client configuration file C:\\Progra~1\\wfavvid\\tomcat_appadmin\\\conf\IPPAClient.properties. 2010-09-22 18:23:03,116 INFO IPPA0002 Language en country US. 2010-09-22 18:23:03,147 INFO IPPA IPPA client locale en_US. 2010-09-22 20:33:41,919 INFO IPPA0003 BIPPA service IOR IOR:01001c0049444c3a73636970686f6e65786d6c2f495050415376723a312e3100610102000c0031302e31302e3231302e350082e60c004c99a0e60d2b0001020008000154544101001c0001000100010001000100010509010100010009010100. 2010-09-22 20:33:42,153 INFO IPPA0005 CTI type 2 used. 2010-09-22 20:33:42,700 WARN IPPA3008 IP phone address 192.168.10.69: IPPA client encountered an error 56:Invalid agent password specified. while making LoginAgent call to BIPPA service. 2010-09-22 20:33:57,138 WARN IPPA3008 IP phone address 192.168.10.69: IPPA client encountered an error 56:Invalid agent password specified. while making LoginAgent call to BIPPA service. 2010-09-22 20:34:49,872 WARN IPPA3008 IP phone address 192.168.10.69: IPPA client encountered an error 56:Invalid agent password specified. while making LoginAgent call to BIPPA service. 2010-09-22 20:36:34,076 WARN IPPA3008 IP phone address 192.168.10.69: IPPA client encountered an error 56:Invalid agent password specified. while making LoginAgent call to BIPPA service. 2010-09-22 20:45:40,733 WARN IPPA3008 IP phone address 10.10.200.2: IPPA client encountered an error 56:Invalid agent password specified. while making LoginAgent call to BIPPA service. Regards Amit --- On Mon, 9/13/10, Tam Nhu tamnhu...@gmail.com wrote: From: Tam Nhu tamnhu...@gmail.com Subject: Re: [OSL | CCIE_Voice] UCCX 7.0(1) Error - Cannot connect to the IPPhone Agent service To: Amit Batra batraji...@yahoo.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Monday, September 13, 2010, 8:28 PM Hi Amit, It's great to hear that you have a uccx vmware running. Do you have any issue with IPPA login with single button? If you have it setup and ready, then please try to login from the phone and then go to uccx server and open the IPPClientxxx.log (the latest one), in C:\Progra~1\wfavvid\tomcat_appadmin\logs folder, and see if you have any error; if not, then please paste the last output of the log. I believed that I have a corrupted vmware image, so I am still searching for another vmware image. If you still remember where you get your vmware image from, that would help also. Thanks. TN. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCCX 7.0(1) Error - Cannot connect to the IPPhone Agent service
Hi Tim, Could you throw some light now which services you have restarted ??? I have restarted my servers.. CAD with 7960 and CIPC working fine... But when i use IPPA service using the phone service...it doesnt work.. Shows password invalid on the phone screen .. regards Amit --- On Tue, 9/21/10, Tam Nhu tamnhu...@gmail.com wrote: From: Tam Nhu tamnhu...@gmail.com Subject: Re: [OSL | CCIE_Voice] UCCX 7.0(1) Error - Cannot connect to the IPPhone Agent service To: Amit Batra batraji...@yahoo.com Cc: osl osl ccie_voice@onlinestudylist.com Date: Tuesday, September 21, 2010, 8:31 PM I finally resolved the issue with VMware. I just needed to run the post-install and corrected the LDAP and Desktop Sync services; everything is running good after that. Thanks all for helps. TN. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCCX 7.0(1) Error - Cannot connect to the IPPhone Agent service
Hi Goran, Should i call you a legend or you will prefer something better , bigger than that... Mate you that was exactlly the case...I change from PWD to Pwd...it worked like a charm.. Tons of thanks mate.. learnt a lesson today Thanks again Regards Amit --- On Wed, 9/22/10, Goran Selthofer seltho...@gmail.com wrote: From: Goran Selthofer seltho...@gmail.com Subject: Re: [OSL | CCIE_Voice] UCCX 7.0(1) Error - Cannot connect to the IPPhone Agent service To: amit batra batraji...@yahoo.com Cc: Tam Nhu tamnhu...@gmail.com, osl osl ccie_voice@onlinestudylist.com Date: Wednesday, September 22, 2010, 2:51 PM Amit, logically, if CAD is working fine and you have issue only with IPPA with password and the password is for sure correct then it only can be one problem - that is that you are sending this pass inside wrongly configured parameter on CUCM ip phone services. those parameters are very case sensitive. so in example, maybe you configured parameter for password to be PWD instead of Pwd in CUCM? check that... cheers! G. On Wed, Sep 22, 2010 at 11:08 AM, amit batra batraji...@yahoo.com wrote: Hi Tim, Could you throw some light now which services you have restarted ??? I have restarted my servers.. CAD with 7960 and CIPC working fine... But when i use IPPA service using the phone service...it doesnt work.. Shows password invalid on the phone screen .. regards Amit --- On Tue, 9/21/10, Tam Nhu tamnhu...@gmail.com wrote: From: Tam Nhu tamnhu...@gmail.com Subject: Re: [OSL | CCIE_Voice] UCCX 7.0(1) Error - Cannot connect to the IPPhone Agent service To: Amit Batra batraji...@yahoo.com Cc: osl osl ccie_voice@onlinestudylist.com Date: Tuesday, September 21, 2010, 8:31 PM I finally resolved the issue with VMware. I just needed to run the post-install and corrected the LDAP and Desktop Sync services; everything is running good after that. Thanks all for helps. TN. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] UCCX RMON setup
Hi GUys, I am trying to setup remote monitoring script. i am using the default RMON script in the application application management. . The directory number i have setup for this is 2123945400 When i dial this number from PSTN phone its says Welcome to remote monitoring.. Please enter your username and pin I have defined a user under the ToolsUser management and assigned reporting capabilities... When i enter the username and pin (( Defined in the CUCM).. UCCX says its incorrect.. Can anyone throw some light on whats wrong ?? If any more info is required please feel free to ask.. regards Amit ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SRST to VM - redirecting number lost
Hi Buddy ,, could you please share what helped you to fixed the issue??? Regards Amit Sent from my iPhone On 19/09/2010, at 7:41 PM, vcciev vcc...@gmail.com wrote: OK, resolved ___ 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] Missed call configuration on CME
Hi Narayan I think you are missing translation profile to add 9 infront of the ANI coming from pstn... The profile can be attached to either voice port or dial-peer used for incoming calls in the incoming direction..with In translation profile calling number... Prefix 9 for subscrber calls...the prefix will be different based on your dialling habbits.. I will send u a sample config... Regards Amit Sent from my iPhone On 17/09/2010, at 6:37 AM, narayan sarma narayan.na...@hotmail.com wrote: Hi Vik Ryan, I am Narayan. This is my first posting on OSL. I am stuck in an config. I configured an CME router everything is working properly. But the problem is that when there is a missed call. I am not able to call the missed call number directly. I have to use 9 then the number. Below is the config. CME#sh run Building configuration... ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname CME ! boot-start-marker boot-end-marker ! card type e1 0 0 no logging console enable secret 5 $1$tocv$mkXQADZJ8hs.M8XP7NN.1/ ! no aaa new-model clock timezone GMT 3 clock summer-time GMT recurring network-clock-participate wic 0 dot11 syslog no ip dhcp use vrf connected ip dhcp excluded-address 10.10.10.1 10.10.10.10 ip dhcp excluded-address 10.10.10.250 10.10.10.254 ! ip dhcp pool phone network 10.10.10.0 255.255.255.0 default-router 10.10.10.254 option 150 ip 10.10.10.254 ! ! ip cef ! ! no ip domain lookup multilink bundle-name authenticated ! isdn switch-type primary-net5 voice-card 0 no dspfarm dsp services dspfarm ! ! ! ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip supplementary-service h450.12 no supplementary-service sip refer h323 sip registrar server expires max 3600 min 3600 ! ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g711alaw codec preference 3 g729br8 ! ! ! ! ! ! voice class custom-cptone jointone dualtone conference frequency 600 900 cadence 300 150 300 100 300 50 ! ! voice class custom-cptone leavetone dualtone conference frequency 400 800 cadence 400 50 200 50 200 50 ! ! ! ! ! ! ! ! ! ! ! ! ! archive log config hidekeys ! ! controller E1 0/0/0 framing NO-CRC4 pri-group timeslots 1-21 ! ! ! ! ! interface GigabitEthernet0/0 ip address x.x.x.x 255.255.255.248 ip virtual-reassembly duplex auto speed auto media-type rj45 ! interface GigabitEthernet0/1 ip address 10.10.10.254 255.255.255.0 ip nat inside ip virtual-reassembly duplex auto speed auto media-type rj45 ! interface Serial0/0/0:15 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn incoming-voice voice no cdp enable ! ip forward-protocol nd ip route 172.16.0.0 255.255.255.0 10.10.10.1 ip route 172.16.1.0 255.255.255.0 10.10.10.1 ip route 192.168.2.0 255.255.255.0 10.10.10.1 ip route 192.168.3.0 255.255.255.0 10.10.10.1 ! ! ip http server ip http authentication local no ip http secure-server ip http path flash: ip nat inside source list NAT interface GigabitEthernet0/0 overload ! ip access-list standard NAT permit 192.168.2.0 0.0.0.255 permit 172.16.0.0 0.0.0.255 permit 172.16.1.0 0.0.0.255 ! access-list 10 permit 192.168.2.0 0.0.0.255 access-list 10 permit 192.168.3.0 0.0.0.255 access-list 10 permit 10.10.10.0 0.0.0.255 ! ! tftp-server flash:P0030702T023.loads tftp-server flash:P0030702T023.sb2 tftp-server flash:P0030702T023.sbn tftp-server flash:P0030702T023.bin tftp-server flash:Analog1.raw tftp-server flash:Analog2.raw tftp-server flash:AreYouThere.raw tftp-server flash:AreYouTheref.raw tftp-server flash:Bass.raw tftp-server flash:CallBack.raw tftp-server flash:Classic1.raw tftp-server flash:Classic2.raw tftp-server flash:ClockShop.raw tftp-server flash:Drums1.raw tftp-server flash:Drums2.raw tftp-server flash:FilmScore.raw tftp-server flash:HarpSynth.raw tftp-server flash:Jamaica.raw tftp-server flash:KotoEffect.raw tftp-server flash:MusicBox.raw tftp-server flash:RingList.xml tftp-server flash:DistinctiveRingList.xml tftp-server flash:Piano1.raw tftp-server flash:Piano2.raw tftp-server flash:Pop.raw tftp-server flash:Pulse1.raw tftp-server flash:Ring1.raw tftp-server flash:Ring2.raw tftp-server flash:Ring3.raw tftp-server flash:Ring4.raw tftp-server flash:Ring5.raw tftp-server flash:Ring6.raw tftp-server flash:Ring7.raw tftp-server flash:Sax1.raw tftp-server flash:Sax2.raw tftp-server flash:Chime.raw tftp-server flash:Vibe.raw tftp-server flash:apps42.8-3-1-22.sbn tftp-server flash:cnu42.8-3-1-22.sbn tftp-server flash:cvm42sccp.8-3-1-22.sbn tftp-server flash:dsp42.8-3-1-22.sbn tftp-server flash:jar42sccp.8-3-1-22.sbn tftp-server flash:SCCP42.8-3-2S.loads tftp-server flash:term42.default.loads tftp-server flash:term62.default.loads ! control-plane ! ! ! voice-port 0/0/0:15 ! voice-port 0/1/0 ! voice-port 0/1/1 ! voice-port 0/1/2 ! voice-port 0/1/3 !
Re: [OSL | CCIE_Voice] CHALLENGES WITH CUE ! ! !
Hi I am not an unity express expert. But I think this is after message action thing.. Should be set to hang up... In Unity connetion it's possible. I don't know how to do this in express.. I hope this helps Regards Amit Sent from my iPhone On 13/09/2010, at 6:57 PM, Pithog Oil pithog...@yahoo.com wrote: This is what i mean, when i call ext 3003 from ext 3002 , and extension 3003 refused to pick up untill the call times out then i get transfered to voice mail, i leave a message for 3003 , then after leaving the message , ext 3002 tells me wait while i transfer your call , then 3003 starts ringing again for the second time. --- On Mon, 9/13/10, cisco voip voip.ccieci...@gmail.com wrote: From: cisco voip voip.ccieci...@gmail.com Subject: Re: [OSL | CCIE_Voice] CHALLENGES WITH CUE ! ! ! To: Pithog Oil pithog...@yahoo.com Cc: ccie_voice@onlinestudylist.com Date: Monday, September 13, 2010, 5:02 AM WAIT WHILE I TRANSFER YOUR CALL is transfer greeting and you don't transfer a call to send the message. Can you please explain how are you sending the message On Mon, Sep 13, 2010 at 10:18 AM, Pithog Oil pithog...@yahoo.com wrote: Hi Experts I configured CUE and it works properly, except for this behaviour which i dont think is the normal behaviour, i get this after sending a voice mail, WAIT WHILE I TRANSFER YOUR CALL then suddenly the phone that timed out without picking the call starts ringing. whearas my Unity connection does not have this behaviour. Also for VPIM my message was sent on CUE agent only to see them return back to my phone after a few minutes, However from Cisco Unity i could not get messages sent accros, i am never prompted to enter my location ID, i know i am missing a critical bit, i need an expert to lighten me up on how to troubleshoot VPIM effectively. Thanks in anticipation ___ 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] UCCX 7.0(1) Error - Cannot connect to the IPPhone Agent service
Hi Tam I have Uccx running on vmware.. Please let me know if you need any information.. Regards Amit Sent from my iPhone On 14/09/2010, at 12:43 AM, Tam Nhu tamnhu...@gmail.com wrote: One more thing, though, before I spend hell out of my time to troubleshoot this. If anyone has your own UCCX server, could you please look at the directory below and see if you can find the file IPPAServerRT.cfg in the tomcat_appadmin folder. Based on the error when I tried to login on the phone, this file C:\Progra~1\wfavvid\tomcat_appadmin\IPPAServerRT.cfg is missing. And based on my research, this file wasn't created or written in the directory when CAD installed. I am confusing a little bit here since this file, CAD, and Tomcat related issues are all applied to UCCE, not UCCX. Do I need CAD for UCCX? It probably that the VMware image I used to setup the UCCX server is corrupted. So if someone please give me a sight based on your UCCX server, it would be very appreciated. Thanks, TN. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCCX 7.0(1) Error - Cannot connect to the IPPhone Agent service
Hi Tam I got the original Uccx installation media from my work. I converted into iso and uploaded on my vmware.. I didn't download it from anywhere.. Regards Amit Sent from my iPhone On 14/09/2010, at 2:58 AM, Tam Nhu tamnhu...@gmail.com wrote: Hi Amit, It's great to hear that you have a uccx vmware running. Do you have any issue with IPPA login with single button? If you have it setup and ready, then please try to login from the phone and then go to uccx server and open the IPPClientxxx.log (the latest one), in C:\Progra~1\wfavvid\tomcat_appadmin\logs folder, and see if you have any error; if not, then please paste the last output of the log. I believed that I have a corrupted vmware image, so I am still searching for another vmware image. If you still remember where you get your vmware image from, that would help also. Thanks. TN. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Unity Connection Phone view
Hi Guys, Thanks for your reply...Yes it all worked well in the end Thanks to all of you regards Amit --- On Sun, 9/12/10, Miron Kobelski findko...@gmail.com wrote: From: Miron Kobelski findko...@gmail.com Subject: Re: [OSL | CCIE_Voice] Unity Connection Phone view To: Tam Nhu tamnhu...@gmail.com Cc: ccie_voice@onlinestudylist.com Date: Sunday, September 12, 2010, 5:52 PM yes, that's the correct digit sequence. my mistake, sorry. -- Sent from my mobile device. On 11 Sep 2010 22:56, Tam Nhu tamnhu...@gmail.com wrote: Hi Amit, Make sure the CTI user you created is an Application User and has CCM Admin User permission. To test it, call in VM, if everything is set up properly, then it should give you option 5 'To Find Message', and then option 4 for 'Display All Msg'. TN. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -Inline Attachment Follows- ___ 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] CME Hunt group question
Hi Guys, I am testing hunt group configuration on CME.. Following is my routers config.. ephone-hunt 1 sequential pilot 3100 list 3002 ephone-hunt 2 sequential pilot 3200 list 3001 This configuration works fine... Now what i wanna do is that ... If call comes for 32143100 (Hunt pilot 3100) it should go to 3002 first. If 3002 is unavailable or doesnt answer then it goes to hunt pilot 3200 (Second hunt group) I tried this and it doesnt work.. Am i missing anything ??? ephone-hunt 1 sequential pilot 3100 list 3002 Final 3200 ephone-hunt 2 sequential pilot 3200 list 3001 Thanks Amit ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Unity Connection Phone view
Hi Guys, I have CUCM 7.0.1 and Unity connection 7.0.1..I am trying to setup phone view feature of Unity connection..Following the document i have done all the steps.. I couldnt find any supporting document and wanna know on how to use this feature...I wanna see this working on my phones.. 1) Created CTI user.. All phones associated.. 2) On Unity, telephony-integration enable phone view and put CTI username and password. 3) in user profilePhone menuenable message locaterenable voice recognition.and enable phone view... Am i missing anything ??? If all above is correct , how to access this feature on cisco 7940,7941,7960 IP phones.. Thanks ... Regards ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] ccie voice home lab
Hi Tariq
You will need Dte-Dce cables. We usually call them back to back cables.
They come in different lengths.. 1 feet 3 feet 6 feet or even longer. Buy
whatever suits ur need..
I hope this helps
Regards
Amit
Sent from my iPhone
On 1/08/2010, at 12:59 AM, Tariq Khan tari...@live.com wrote:
Dear all,
I am setting ccie voice lab at home, i have bought 3 1760 routers and 1 2811
router and 1 2523 router and 1 3550 switch and some wic's and vwic's.
My topology will be like 1 frame relay swith (2523). 1 PSTN Router (1760) and
remaining 3 routers as HQ, BR1 BR2.
1) i want to connect all the routers to 2523 router which i want to make a
frame-relay switch,
{my question is what are the cables through which i should connect my
remaining 4 routers to this routers because it came with serial interfaces}
I would really appreciate if someone can help me in connecting all these
routers with the correct vwic's, wic's and cables.
Thanks Regards
Tariq Khan.
___
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
