Re: [OSL | CCIE_Voice] some calls just ring
Hello , can you please clarify the exact call flow and the behavior in more details ? also Yes deb ccsip mess will tell us about the signalling and the ringback method been used , however i saw that you are changing the invite header and you are changing the media status from sendonly to send recieve , any specific reason for that ? Ash On Thu, Jun 21, 2012 at 9:19 PM, Bill Lake whl...@gmail.com wrote: Hello, Having some trouble and hoping someone has seen this before as I have not been able to reproduced it. Took over this site and trying to resolve some callers getting ring back but never processes at the site. Here are the configurations and I have removed identifiable information. I am thinking I need to debug sip messages but wondering before I do that if anyone has a better idea. Sensitive site as we are taking it over. Router Building configuration... Current configuration : 16282 bytes ! ! Last configuration change at 17:39:38 EST Fri Jul 22 2011 by Administrator ! NVRAM config last updated at 17:40:56 EST Fri Jul 22 2011 by Administrator ! NVRAM config last updated at 17:40:56 EST Fri Jul 22 2011 by Administrator version 15.1 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname 2901 ! boot-start-marker boot system flash:c2900-universalk9-mz.SPA.151-4.M1.bin boot-end-marker ! ! logging buffered 1 ! no aaa new-model clock timezone EDT -5 0 clock summer-time EST recurring ! no ipv6 cef ip source-route ip cef ! ! ! ip dhcp excluded-address 172.16.13.0 172.16.13.10 ip dhcp excluded-address 172.16.13.200 172.16.13.255 ! ip dhcp pool Voice network 172.16.13.0 255.255.255.0 default-router 172.16.13.254 option 150 ip 172.16.13.254 ! ! no ip domain lookup ip domain name CME.local multilink bundle-name authenticated ! ! ! ! ! ! trunk group FXO ! crypto pki token default removal timeout 0 ! voice-card 0 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 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none sip ! voice class codec 1 codec preference 1 g711ulaw ! voice class h323 1 h225 timeout tcp establish 3 h225 display-ie ccm-compatible ! voice class sip-profiles 1 request REINVITE sdp-header Audio-Attribute modify sendonly sendrecv ! ! voice hunt-group 1 parallel list 1301,1302,1303,1304,1305,1306,1310,1311,1312,1320,1321,1322 pilot 7510 ! ! ! ! voice translation-rule 1234 rule 1 /^\*\(13..\)/ /\1/ ! ! voice translation-profile DirectTransfer2VM translate redirect-called 1234 ! ! license udi pid CISCO2901/K9 sn FTX151901AD hw-module ism 0 ! hw-module pvdm 0/0 ! ! ! archive log config logging enable path flash:backup-cfg maximum 14 write-memory username Administrator privilege 15 secret 5 removed username gwashington password 0 removed username jadams password 0 removed username tjefferson password 0 removed username jmadison password 0 removed username jmonroe password 0 removed username jqadams password 0 removed username ajackson password 0 removed username mvanburen password 0 removed username wharrison password 0 removed ! redundancy ! ! ! ! ! ! interface Embedded-Service-Engine0/0 no ip address shutdown ! interface GigabitEthernet0/0 description CONNECTION TO 2960 SWITCH no ip address duplex auto speed auto ! interface GigabitEthernet0/0.1 description DATA VLAN encapsulation dot1Q 1 native ip address 192.168.13.254 255.255.255.0 ip helper-address 192.168.241.76 ! interface GigabitEthernet0/0.2 description VOICE VLAN encapsulation dot1Q 2 ip address 172.16.13.254 255.255.255.0 ! interface ISM0/0 ip unnumbered GigabitEthernet0/0.2 service-module ip address 172.16.13.253 255.255.255.0 !Application: CUE Running on ISM service-module ip default-gateway 172.16.13.254 ! interface GigabitEthernet0/1 no ip address shutdown duplex auto speed auto ! interface ISM0/1 description Internal switch interface connected to Internal Service Module no ip address shutdown ! interface Serial0/0/0 description ** WAN ** bandwidth 1544 ip address removed no fair-queue service-module t1 timeslots 1-24 ! interface Vlan1 no ip address ! ip forward-protocol nd ! ip http server ip http authentication local ip http secure-server ip http path flash: ! ip route 0.0.0.0 0.0.0.0 removed ip route 172.16.13.253 255.255.255.255 ISM0/0 ! ! ! tftp-server flash:apps42.9-1-1TH1-16.sbn tftp-server flash:cnu42.9-1-1TH1-16.sbn tftp-server flash:cvm42sccp.9-1-1TH1-16.sbn
Re: [OSL | CCIE_Voice] Tragic news
I am very sorry to hear this , he was a very nice person , RIP Jeferson , Ash CCIE Voice # 31524 On Fri, Feb 24, 2012 at 10:20 PM, Antonio Dee antonio_...@hotmail.com wrote: very sad news ; Deepest condolensce to Jeferson and family :-( Antonio Dee CCIE RS #25609 Date: Sat, 25 Feb 2012 01:59:03 -0200 From: aedamasc...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Tragic news Hello brothers, I just like to let you know that my friend and study partner, Jeferson Guardia CCIE #28157, has passed away yesterday. He was a skateboarder and day before yesterday, he fell, hitting his head. He didn't want to go to the hospital, because he thought he was ok. According to his father he passed in his sleep due to a blood clot in his brain. This is a tragic moment for all his family and friends. I thought I should share this with you guys because he's been very active here on the list, and we were studying together for the CCIE Voice. He was a great motivator and helped me get out of my personal problems so I could focus on my studies. It's sad how life is, and what shocks everybody the most is that he was only 24 years old (soon to be 25 on March 20th). Mourning, but still on the fight... =( Emanuel Damasceno CCNP Voice ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Cluster Security password for call manger version 4.1
Longtime since i play around 4.x but i guess the CCMPWDChanger Tool or the adminutility is the way . The password is stored in the sql db but i dont have this ccm to brows the table location for you , Check this link and see if it will help : http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a0080557ba5.shtml Ashraf On Tue, Jan 24, 2012 at 11:12 AM, Alpesh Bhakta bhakta.alp...@gmail.com wrote: Hi, Can anybody please help me out For:- 1.How to find out what is my cluster security password for call manager version 4.1 2.How to reset cluster security password for call manager version 4.1 ___ 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] MGCP Registration
the issue you facing is a L2 protocol mismatch issue , The ISDN you using is Qsig so you using Qsig tunneling ? what is your other side switch talking ? and what do you have configured on the CCM config page ? Let me see the debug output please Deb mgcp pack deb isdn q921 Ash On Mon, Jan 16, 2012 at 11:11 AM, Alexander Suhandi alexander.suha...@ag-it.com wrote: Hi mercy, did you try to put the DN number on this particular ISDN? Is the switch-type already correct according to the provider? Salam, Suhandi Sent from Samsung tablet mercy forall mercy_for_...@hotmail.com wrote: this is :- show isdn status \ Global ISDN Switchtype = primary-qsig %Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output may not apply ISDN Serial0/2/0:15 interface dsl 0, interface ISDN Switchtype = primary-qsig Slave side configuration L2 Protocol = Q.921 0x L3 Protocol(s) = CCM MANAGER 0x0003 Layer 1 Status: ACTIVE Layer 2 Status: TEI = 0, Ces = 1, SAPI = 0, State = TEI_ASSIGNED Layer 3 Status: 0 Active Layer 3 Call(s) Active dsl 0 CCBs = 0 The Free Channel Mask: 0x800F Number of L2 Discards = 0, L2 Session ID = 6 Total Allocated ISDN CCBs = 0 Date: Mon, 16 Jan 2012 17:44:32 +0400 Subject: Re: [OSL | CCIE_Voice] MGCP Registration From: datucha...@gmail.com To: mercy_for_...@hotmail.com CC: gogli...@gmail.com; ccie_voice@onlinestudylist.com What does the show isdn status shows up? On Mon, Jan 16, 2012 at 5:35 PM, mercy forall mercy_for_...@hotmail.commailto:mercy_for_...@hotmail.com wrote: Hi, thanks for your support and good link\ this is my GW configuration , also it is connected to other cisco GW as PSTN GW through E1 cross cable sh run : Current configuration : 15381 bytes ! ! version 15.0 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname ! boot-start-marker boot-end-marker ! logging buffered 51200 warnings no aaa new-model network-clock-participate wic 2 ! dot11 syslog ip source-route ! ip cef ! ! no ipv6 cef multilink bundle-name authenticated ! ! ! isdn switch-type primary-qsig ! voice-card 0 ! ! voice rtp send-recv ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip redirect ip2ip h323 sip header-passing no call service stop ! voice class codec 1 codec preference 1 g711ulaw ! voice class custom-cptone dualtone disconnect frequency 425 cadence 250 250 ! ! ! ! http client cache memory pool 15000 http client cache memory file 500 http client connection timeout 60 http client connection idle timeout 10 http client response timeout 30 mrcp client timeout connect 10 mrcp client timeout message 10 mrcp client rtpsetup enable vxml tree memory 500 vxml audioerror vxml version 2.0 ! crypto pki trustpoint TP-self-signed-3307538538 enrollment selfsigned subject-name cn=IOS-Self-Signed-Certificate-3307538538 revocation-check none rsakeypair TP-self-signed-3307538538 ! controller E1 0/2/0 pri-group timeslots 1-4,16 service mgcp ! interface GigabitEthernet0/0 no ip address duplex auto speed auto media-type rj45 ! interface GigabitEthernet0/0.1 encapsulation dot1Q ip address X.X.X.X 255.255.255.0 ! ! interface Serial0/2/0:15 no ip address encapsulation hdlc isdn switch-type primary-qsig isdn incoming-voice voice isdn bind-l3 ccm-manager no cdp enable ! ip forward-protocol nd ! ! ip http server ip http access-class 23 ip http authentication local ip http secure-server ip http timeout-policy idle 60 life 86400 requests 1 ip route 0.0.0.0 0.0.0.0 X.X.X.X ! ! ! control-plane ! call threshold global cpu-5sec low 70 high 85 ! voice-port 0/2/0:15 ! voice-port 0/3/0 ! voice-port 0/3/1 ! ccm-manager switchback immediate ccm-manager fallback-mgcp ccm-manager redundant-host X.X.x.x ccm-manager mgcp ccm-manager config server x.x.x.x ! mgcp mgcp call-agent x.x.x.x service-type mgcp version 1.0 mgcp bind control source-interface GigabitEthernet0/0.1 mgcp bind media source-interface GigabitEthernet0/0.1 ! mgcp profile default ! ! gateway timer receive-rtp 1200 ! sip-ua retry invite 1 retry bye 1 retry cancel 1 timers expires 6 reason-header override ! ! telephony-service max-conferences 12 gain -6 transfer-system full-consult ! thanks From: gogli...@gmail.commailto:gogli...@gmail.com Date: Mon, 16 Jan 2012 13:29:49 +0100 Subject: Re: [OSL | CCIE_Voice] MGCP Registration To: mercy_for_...@hotmail.commailto:mercy_for_...@hotmail.com CC: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Hi, The card is supported:
Re: [OSL | CCIE_Voice] Cant connect to Sub
Hello Errol , This issue usually mean you have incorrect Host/ processNode files in your PUB or the Sub and so the communication is broken between the PUB and the SUB and you have to find what table we have corrupted and fix it , or you have a big timing difference reported by your NTP , can you check what service you have running on both pub and sub specially the A Cisco DB and let me know what you will get . Ash On Mon, Jan 16, 2012 at 5:15 PM, Errol Abrahams eabraham2...@gmail.com wrote: Hi All, I had a problem with my VMWARE Server and I had to rebuilt the system from scratch. I have reloaded PUB,SUB,CUPS,CUC and CUCCX and all virtual addresses are pingable. When I activate the services for the PUB from the Cisco Unified Serviceability screen, it worked. But, when I try to access the SUB from same screen then it displays'Connection to the Server cannot be established(unable to access Remote Node). Has anybody had a problem like this and how can I fix this problem. Your help is appreciated..thnx. Chhers EA ___ 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] QoS question on Workbook2 Lab 10
Hello John , You thinking is correct however as the Packet will traverse over the wan , it will be always subject to get modified by a lot of SW to change specially the DSCP Value and so the QOS SRND recommend to Not trust the packets coming from the wan and so we dont trust in the Branches routers . The right answer for any QOS Question is what is in the SRND and i believe thats why you will find it in the real lap in the desktop Ash On Mon, Jan 16, 2012 at 8:18 PM, John McGaughey (jomcgaug) jomcg...@cisco.com wrote: Hello, In Workbook 2, Lab 10, question 5.2 it asks you to setup MLP LFI between HQ and BR1. In the solution guide it has you use auto qos trust on the HQ side but does not use trust on the BR1 side. The DSG guide says the reason for not using the trust key word is because of the following: Note that we have not done any prior QOS classification/marking on the ESW module therefore we will use class-based marking (no use of the trust keyword when running auto qos). But the phones use the following markings by default. signaling (SCCP or SIP) - CoS 3 / cs3 media (RTP) - CoS 5 / DSCP 46 (EF) Why couldn’t we just use the trust keyword on BR1 as well since the phone is already marking the packets correctly? John ___ 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] RSVP CAC
did you run qos on sc as well ? you know that you have to otherwise you will kill sc sub interface , do show traff and you will see it getting 56 K only , so the solution is to run the QOS on both sub interfaces on HQ and on sc as well Ash On Fri, Jan 13, 2012 at 12:09 AM, study buddy studybudd...@gmail.com wrote: Hi All, For the Lab topology, if I run QoS between HQ SB RSVP CAC between HQ SC set the ip rsvp band to 112 for 4 calls, I can actually make only two calls the third calls get re-routed. Now if I run qos between HQ SC routers as well with a BW of 1536 I can make 3 calls, but the 4th call still fails I get the following RSVP error *Jan 10 12:58:29.883: QoS Module: RESV ERROR received : Remote IP: 142.102.64.254 | Local IP: 142.102.66.254 *Jan 10 12:58:29.883: qos_rsvp_resv_notify_events: errCode 1, errVal 2, errFlag 0, errNode 10.10.112.1 *Jan 10 12:58:29.883: qos_rsvp_remove_reservation: Removing RESV state for CallId : 0xFFC5 Remove Resv: Source (142.102.64.254:17084), Dest (142.102.66.254:17178) Any thoughts on this? TR ___ 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] SIP MWI issues with CUE
Hi , You are binding the Sip to the loopback interface while you using the Vlan400 for the CUE , change this and it should work voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to sip sip bind control source-interface Loopback0 bind media source-interface Loopback0 Ash On Mon, Jan 2, 2012 at 9:54 AM, Rajasekar Shanmugam rajaseka...@gmail.com wrote: Peter - Thanks a lot for your guidance and that made me to realize my config mistake. Actually , the SIP binding was done for a different interface than the telephony-service bound interface. Good catch...:) Raj 2012/1/2 Farkas Péter wormh...@sch.bme.hu SIP stack on CUE has to be given IP address of binded SIP IP address of voice gateway. Try to correct on CUE: ! ccn subsystem sip gateway address 10.10.110.3 ! It may require reset of CUE, as well. Peter - Original Message - From: Rajasekar Shanmugam rajaseka...@gmail.com Date: Monday, January 2, 2012 4:29 pm Subject: [OSL | CCIE_Voice] SIP MWI issues with CUE To: ccie_voice@onlinestudylist.com Experts - I`m running into some issues with the CUE MWI , when working with SRST. I have the required configs using the unsolicited notify for the MWI. Attached my configs the ccsip debug output. Not sure , wher I`m going wrong. Please help. -- Raj ___ 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 -- Raj ___ 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] Cluster Security password
Happy new Year Asad and everyone, Here is summary steps of how to do that : 1. Log in to the system with the following username and password: Username: pwrecovery Password: pwreset The Welcome to platform password reset window displays. 2. Press any key to continue. 3. If you have a CD or DVD in the disk drive, remove it now. 4. Press any key to continue. The system tests to ensure that you have removed the CD or DVD from the disk drive. 5. Insert a valid CD or DVD into the disk drive. For this test, you must use a data CD, not a music CD. The system tests to ensure that you have inserted the disk. 6. After the system verifies that you have inserted the disk, you get prompted to enter one of the following options to continue: Enter a to reset the administrator password. Enter s to reset the security password. Enter q to quit. 7. Enter a new password of the type that you chose. 8. Reenter the new password. The password must contain at least 6 characters. The system checks the new password for strength. If the password does not pass the strength check, you get prompted to enter a new password. 9. After the system verifies the strength of the new password, the password gets reset, and you get prompted to press any key to exit the password reset utility. if you still facing error after this procedures , collect cluster manager service traces and send them over to me i will check it for you Ash On Sun, Jan 1, 2012 at 10:31 PM, Cisco Nut rafayc...@gmail.com wrote: Asad Checkout the following link http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/cucos/7_1_2/cucos/iptpch2.html#wp1044244 On Sun, Jan 1, 2012 at 6:42 AM, Asad Yasin asad4nt...@gmail.com wrote: Any body know how to change the cluster password that is used on subscriber to join to publisher. Thanks 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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] lab/6 MGCP TS got the lab in Dubai really hard luck
Thank you so much , as suspected , this is a fake email ID and this is a way to sell your LAB 6 , in regard of my credibility , i don't need someone like you to tell if i can get the marks or no , alot of people know me and also Cisco itself know me , If anyone interested of the Trash you trying to sell its his problem Now , i did what i wanted and now this way is too old , you go to dubai and got lab 6 and it was hard and all this stuff ...Once again thank you ..Find another way ! Ashraf On Mon, Dec 12, 2011 at 11:48 AM, Marksnap Marky markysna...@yahoo.com wrote: Ashraf If u will write this in lab u will get 0 hehehehehe :) thanks for yr suggestions but i am nt interested :) If anyone has proper answers ping me :) From: Ashraf Ayyash ash.ayy...@gmail.com To: Marksnap Marky markysna...@yahoo.com Cc: Ray jonha...@yahoo.com; ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Monday, December 12, 2011 12:09 PM Subject: Re: [OSL | CCIE_Voice] lab/6 MGCP TS got the lab in Dubai really hard luck Hello , I don't appreciate your unprofessional /NDA violation you bring to the alias and i am not replaying to this email for you , i am replaying because i know you scare so many people who about getting the exam soon ...Thank you for this and feel free to Ignore my email , Now , for the people who maybe interested , if you will have to troubleshoot audio issue with MGCP , you have to know the following : 1- from the debug mgcp pack check the source and the port and the codec and the media type from the SDP in the CRCX and the MDCX and the 200 OK , you have to know where to look and what every character in the SDP means , refer to email sent today in the alias about this and also you can always check the RFC . from there you have to find out if this is a connectivity issue ( ping the ip from the C parameter with the source of the correct IP you binding the MGCP in and make sure there is no Binding/connectivity issue , if so then you good to explain what is going on , 2- in the LAB version you may Hit CSCsy10653 which can cause No way audio issue and you have to know the Symmetrical Support for MGCP Call Based : http://www.cisco.com/en/US/docs/ios/12_4t/12_4t11/ht_6974s.html and of course the solution of the defect from the release notes : CSCsy10653 Symptoms: Calls on an MGCP gateway negotiating the g729br8 codec may fail to have audio in one or both directions. Conditions: This occurs on MGCP gateways with the fix for CSCsu66759 when the g729br8 codec is being negotiated. Workaround: Any of the following will be sufficient to get around this issue: 1. Configure the gateway for static payload type using the following commands on the gateway: mgcp behavior g729-variants static-pt mgcp behavior dynamically-change-codec-pt disable 2. Disable g729br8 from being negotiated for this call. If CUCM is involved, this is done with the service parameter Strip G.729 Annex B (Silence Suppression) from Capabilities. 3. Use a Cisco IOS code on the gateway which does not contain the fix for CSCsu66759 (Cisco IOS Release 12.4(22)T and below). http://www.cisco.com/en/US/docs/ios/12_4t/release/notes/124TCAVS1.html Ash On Sun, Dec 11, 2011 at 3:08 PM, Ray jonha...@yahoo.com wrote: check out this link,, it may help ... http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a008009484b.shtml From: Marksnap Marky markysna...@yahoo.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Sunday, December 11, 2011 12:03 PM Subject: [OSL | CCIE_Voice] lab/6 MGCP TS got the lab in Dubai really hard luck Can anyone guide how to solve lab 6 MGCP TS 3.4 MGCP Troubleshooting Management has confirmed that there are instances of one way audio from outbound calls made from HQ phones. Please provide the appropriate debug to verify whether or not One way audio instances are prevalent for HQ Phones. Only provide the appropriate debug instance together with an explanation highlighting your response to Management in no less than 50 words in a text file titled MGCP.txt on the User PC’s desktop. can anyone say how to solve this!!! shit again missed 3 time thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE
Re: [OSL | CCIE_Voice] MGCP Parameters Documentation
Hello Joe, I don't think there is any Doc that explain the MGCP SDP parameters as this is usually a TAC level of analysis , However if you have any specific Question about the MGCP parameter i will may Help you Ash On Sat, Dec 10, 2011 at 2:42 PM, AJ BG ciscoie2...@gmail.com wrote: Friends, Is there any Cisco document that explains MGCP Parameters, similar to the table bellow? *MGCP Parameters* *Code* *Parameter Name* A Capabilities B BearerInformation C CallId D DigitMap E ReasonCode ES EventStates F RequestedInfo I ConnectionId I2 SecondConnectionId K ResponseAck L LocalConnectionOptions LC LocalConnection Descriptor M ConnectionMode N NotifiedEntity O ObservedEvents P ConnectionParameters Q QuarantineHandling R RequestedEvents RC RemoteConnectionDescriptor RD RestartDelay RM RestartMethod S SignalRequests T DetectEvents X RequestIdentifier Z SpecificEndpointID Z2 Second Endpoint ID I know that the RFC 2705 explains it. But I would like to find a reference in Cisco documentation website. So it will be accessible during the lab as well. thanks, Joe ___ 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] paging with CallManager 7
Hello , i have published this Info when i was in the TAC as i got email of features included in the 8.6 CCM , the best person to ask about this is your account manager , he can talk to the BU about this and get back to you with better answer Ashraf On Sun, Dec 11, 2011 at 8:48 PM, Adil Shaikh adil.sha...@gmail.com wrote: Hi Ashraf, 8.6(2) or 8.6(2a) release note does not have paging feature information. On cisco site, Features and services guide is still version 8.6.1, so, can't check whether paging is included in 8.6.2 or 8.6.2a. I have not got CUCM 8.6.2, so can't check it. So, has santa claus arrived and we have no hint of it? thanks -adil On Tue, Aug 30, 2011 at 4:29 PM, Ashraf Ayyash ash.ayy...@gmail.comwrote: heheh , santan claus will work for Cisco this year to create the Paging feature !! now to confirm this feature with some more details , Paging feature will be introduced in 8.6(2) version of CCM which will be release before the end of this Year , in addition , 8.6(2) will have more cool features , one worth to mention is Redirecting Number Xformation :) Best regards Ash On Tue, Aug 30, 2011 at 4:39 AM, Roger Carpio roger.car...@gmail.comwrote: Don't you worry boys... I know if we behave Santa Claus will program it for us... LOL On Mon, Aug 29, 2011 at 12:56 PM, Ashraf Ayyash ash.ayy...@gmail.comwrote: the release still coming over in the few next weeks,the release notes for that version is not yet published . Ash On Mon, Aug 29, 2011 at 7:19 PM, Mark Reed marklr...@gmail.com wrote: Where did you see that? I looked at the 8.6 release notes and don't see that feature. On Mon, Aug 29, 2011 at 3:20 AM, Ashraf Ayyash ash.ayy...@gmail.comwrote: Hello Erwan , Paging Feature will be available on the latest 8.6 CCM Version , Ash On Mon, Aug 29, 2011 at 8:38 AM, Erwan Erwan e_er...@yahoo.comwrote: hi all, does call manager 7 have paging feature we can use ? for basic paging tks ___ 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 -- Thanks, Mark L Reed Home: 260-637-1585 ___ 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 -- .. . . _7___|___|_|_|adil.sha...@gmail.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] CUE misbehaving
what is the status of this module ? ser ser 0/0 status , reset ..play with it a lil bit , make sure you have the CUE SW installed Ash On Sun, Dec 11, 2011 at 6:30 PM, Randall Crumm rrcr...@yahoo.com wrote: HI, I have an issue with my CUE. I have the configuration on the SC rtr (see below). When I enter ser ser 0/0 ses I just get blank ... I hit ctl+alt+6 x and get back to the sc rtr cli If I hit enter again I get : SiteC-RTR# [Resuming connection 1 to 10.10.115.1 ... ] This is blank space... interface Loopback1 ip address 10.10.115.1 255.255.255.0 ip ospf network point-to-point ! interface Service-Engine0/0 ip unnumbered Loopback1 service-module ip address 10.10.115.2 255.255.255.0 service-module ip default-gateway 10.10.115.1 ip route 10.10.115.2 255.255.255.255 Service-Engine0/0 Thanks, Randall ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] lab/6 MGCP TS got the lab in Dubai really hard luck
Hello , I don't appreciate your unprofessional /NDA violation you bring to the alias and i am not replaying to this email for you , i am replaying because i know you scare so many people who about getting the exam soon ...Thank you for this and feel free to Ignore my email , Now , for the people who maybe interested , if you will have to troubleshoot audio issue with MGCP , you have to know the following : 1- from the debug mgcp pack check the source and the port and the codec and the media type from the SDP in the CRCX and the MDCX and the 200 OK , you have to know where to look and what every character in the SDP means , refer to email sent today in the alias about this and also you can always check the RFC . from there you have to find out if this is a connectivity issue ( ping the ip from the C parameter with the source of the correct IP you binding the MGCP in and make sure there is no Binding/connectivity issue , if so then you good to explain what is going on , 2- in the LAB version you may Hit CSCsy10653 which can cause No way audio issue and you have to know the Symmetrical Support for MGCP Call Based : http://www.cisco.com/en/US/docs/ios/12_4t/12_4t11/ht_6974s.html and of course the solution of the defect from the release notes : CSCsy10653 Symptoms: Calls on an MGCP gateway negotiating the g729br8 codec may fail to have audio in one or both directions. Conditions: This occurs on MGCP gateways with the fix for CSCsu66759 when the g729br8 codec is being negotiated. Workaround: Any of the following will be sufficient to get around this issue: 1. Configure the gateway for static payload type using the following commands on the gateway: mgcp behavior g729-variants static-pt mgcp behavior dynamically-change-codec-pt disable 2. Disable g729br8 from being negotiated for this call. If CUCM is involved, this is done with the service parameter Strip G.729 Annex B (Silence Suppression) from Capabilities. 3. Use a Cisco IOS code on the gateway which does not contain the fix for CSCsu66759 (Cisco IOS Release 12.4(22)T and below). http://www.cisco.com/en/US/docs/ios/12_4t/release/notes/124TCAVS1.html Ash On Sun, Dec 11, 2011 at 3:08 PM, Ray jonha...@yahoo.com wrote: check out this link,, it may help ... http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a008009484b.shtml From: Marksnap Marky markysna...@yahoo.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Sunday, December 11, 2011 12:03 PM Subject: [OSL | CCIE_Voice] lab/6 MGCP TS got the lab in Dubai really hard luck Can anyone guide how to solve lab 6 MGCP TS 3.4 MGCP Troubleshooting Management has confirmed that there are instances of one way audio from outbound calls made from HQ phones. Please provide the appropriate debug to verify whether or not One way audio instances are prevalent for HQ Phones. Only provide the appropriate debug instance together with an explanation highlighting your response to Management in no less than 50 words in a text file titled MGCP.txt on the User PC’s desktop. can anyone say how to solve this!!! shit again missed 3 time thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ 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] UCCX on VMWARE problems.
do you have the License uploaded to the CCX , this can happen when you have no license Ash On Thu, Dec 8, 2011 at 12:20 PM, ccielabrat ccielab...@gmail.com wrote: I just wanted to do some testing on UCCX so I booted a vmware image of UCCX that I've used before. It's a fresh install with no integration. When I log in, it says the JTAPI is out of sync. I've fixed the JTAPI problem related to moving C:\windows\java files to c:\winnt\java It wants me to rerun jtapi sync, but the menus on the UCCX page page will not display correctly. Do dropdown menus appear. Can anyone help? ___ 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] Silent Moh File
You can get it from the UCCX server in the prompt file , Ash On Thu, Dec 8, 2011 at 1:25 PM, William Affeldt william.affe...@yahoo.com wrote: Does anyone have a silent Moh file already created that I can use? ___ 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] UCCX on VMWARE problems.
What do you mean ? Where are you at exactly in the CCX installation ? Screenshot? Ash On Thursday, December 8, 2011, ccielabrat ccielab...@gmail.com wrote: I can't get to the point to upload the license. On Thu, Dec 8, 2011 at 2:07 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote: do you have the License uploaded to the CCX , this can happen when you have no license Ash On Thu, Dec 8, 2011 at 12:20 PM, ccielabrat ccielab...@gmail.com wrote: I just wanted to do some testing on UCCX so I booted a vmware image of UCCX that I've used before. It's a fresh install with no integration. When I log in, it says the JTAPI is out of sync. I've fixed the JTAPI problem related to moving C:\windows\java files to c:\winnt\java It wants me to rerun jtapi sync, but the menus on the UCCX page page will not display correctly. Do dropdown menus appear. Can anyone help? ___ 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] UCCX on VMWARE problems.
Good To know you fix it , and its always my pleasure to help Ash On Thu, Dec 8, 2011 at 7:52 PM, ccielab...@gmail.com wrote: Hi Ash, I stand corrected, I had uploaded the license file. I was at the point right after the initial setup completes and prompts you to close your browser. At that point, it presented the jtapi error. I got it working though. I ran setup /x from the UCCX media image and uninstalled UCCX. I then re-ran setup and have a clean UCCX server to mess with. Thank you as always for jumping in to help. On Thu, Dec 8, 2011 at 5:02 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote: What do you mean ? Where are you at exactly in the CCX installation ? Screenshot? Ash On Thursday, December 8, 2011, ccielabrat ccielab...@gmail.com wrote: I can't get to the point to upload the license. On Thu, Dec 8, 2011 at 2:07 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote: do you have the License uploaded to the CCX , this can happen when you have no license Ash On Thu, Dec 8, 2011 at 12:20 PM, ccielabrat ccielab...@gmail.com wrote: I just wanted to do some testing on UCCX so I booted a vmware image of UCCX that I've used before. It's a fresh install with no integration. When I log in, it says the JTAPI is out of sync. I've fixed the JTAPI problem related to moving C:\windows\java files to c:\winnt\java It wants me to rerun jtapi sync, but the menus on the UCCX page page will not display correctly. Do dropdown menus appear. Can anyone help? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Problem with Multicast MOH from router flash
The show ccm music will not show you any output for Voip calls , this is used when you call from the PSTN to the BR1 and place the call on hold ( in other word when the DSP get involved as the ccm music command is to make the DSP injecting the MOH stream into the Isdn Call ) so this is normal , I think you Missing the Voice class codec under the incoming Voip Dial-peer on the BR1 router , you need it as the Call will be swapped to G711u when you will place it on hold , make sure you have it , Also what do you hear when you Put the PSTN call on hold ? in this case you can use the show ccm music to see what codec you are using , Ash On Thu, Dec 1, 2011 at 1:24 AM, Rynard Coetzee rynard.coet...@bytes.co.za wrote: But I am streaming the MoH from the branch router ,so it should not be using a dial-peer ? Or does it still use the dial-peer as it thinks it is streaming from the CUCM server ? From: William Affeldt [mailto:william.affe...@yahoo.com] Sent: 30 November 2011 08:46 PM To: Rynard Coetzee Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Problem with Multicast MOH from router flash Make sure that you configured the codec under the dial peers on BR1. They use g729 by default. Sounds like the dial-peers to call manager from BR1 don't have a voice class codec or are a hard set codec configured. Sent from my iPhone On Nov 30, 2011, at 8:12 AM, Rynard Coetzee rynard.coet...@bytes.co.za wrote: Hi guys I have been struggling with this problem for most of the day now ,and I don`t know what else I can check. The MoH works if I call from HQ branch to BR1 ,but when I call from the PSTN to BR1 I get tone on hold. I have the relevant config on my BR1 router Ccm-manager music-on-hold Telephony service Moh music-on-hold.au Multicast moh 239.1.1.1 port 16384 route 10.10.110.2 (loopback) 10.10.201.1 (Voice Int) I also configured separate DP for MOH server in CUCM with a G711 only region. I have the MOH setting on the audio source ,server and MRG set. Another strange thing that I see is when I call from HQ phone to BR1 phone and I put BR1 phone on hold I hear the MOH stream ,but when I do show ccm-manager music-on-hold it shows that there are 0 active multicast sessions ? Any ideas ,because I have run out … ? Regards Rynard ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Adding second language to CUE
This mean you are installing the Wrong Language files , or you missing on critical file , can you please paste what you have in the FTP directory root ? Ash On Wed, Nov 30, 2011 at 1:03 PM, ccielabrat ccielab...@gmail.com wrote: I'm trying to add a second language to an AIM-CUE. I use the command software install add url ftp://x.x.x.x/xyz.pkg and it seems to run without a problem but when it finishes processing the file, I get the follow message : Language add-ons found on the system (1): Installed SKU Name (version) -- * ENU CUE Voicemail US English (7.0.6) Maximum allowed language add-ons (=1) already installed. You can use software uninstall to remove add-ons. ui_install scripts executed successfully. The issue is if I run Show software licenses , it indicates a max of 2 languages are allowed. CUE# sho software licenses Installed license files: - voicemail_lic.sig : 12 MAILBOX LICENSE Core: - Application mode: CCME - Total usable system ports: 6 Voicemail/Auto Attendant: - Max system mailbox capacity time: 840 - 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: 2 - Max enabled languages: 2 ___ 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] Adding second language to CUE
Hello , why you have tow packages in the root directory ? you have to have the full package of 7.0.6 and the lang pack of GB 7.0.6 ONLY on the root directory , run the installation again and see how it will go Ash On Thu, Dec 1, 2011 at 8:02 AM, ccielabrat ccielab...@gmail.com wrote: Hi Ashraf, See below. Thank you! ftp ls 200 Port command successful 150 Opening data channel for directory list. cue-installer.nm-aim.7.0.1 cue-installer.nm-aim.7.0.6 cue-vm-en_GB-langpack.nm-aim.7.0.6.prt1 cue-vm-full-k9.nm-aim.7.0.1.prt1 cue-vm-full-k9.nm-aim.7.0.6.prt1 cue-vm-installer-k9.nm-aim.7.0.1.prt1 cue-vm-installer-k9.nm-aim.7.0.6.prt1 cue-vm-k9.nm-aim.7.0.1.pkg cue-vm-k9.nm-aim.7.0.1.zip cue-vm-k9.nm-aim.7.0.6.pkg cue-vm-k9.nm-aim.7.0.6.zip cue-vm-k9.nmx.7.1.2.zip cue-vm-langpack.nm-aim.7.0.1.pkg cue-vm-langpack.nm-aim.7.0.6.pkg cue-vm-license_12mbx_ccm_7.0.1.pkg cue-vm-license_12mbx_ccm_7.0.6.pkg cue-vm-license_12mbx_cme_7.0.1.pkg cue-vm-license_12mbx_cme_7.0.6.pkg On Thu, Dec 1, 2011 at 2:57 AM, Ashraf Ayyash ash.ayy...@gmail.com wrote: This mean you are installing the Wrong Language files , or you missing on critical file , can you please paste what you have in the FTP directory root ? Ash On Wed, Nov 30, 2011 at 1:03 PM, ccielabrat ccielab...@gmail.com wrote: I'm trying to add a second language to an AIM-CUE. I use the command software install add url ftp://x.x.x.x/xyz.pkg and it seems to run without a problem but when it finishes processing the file, I get the follow message : Language add-ons found on the system (1): Installed SKU Name (version) -- * ENU CUE Voicemail US English (7.0.6) Maximum allowed language add-ons (=1) already installed. You can use software uninstall to remove add-ons. ui_install scripts executed successfully. The issue is if I run Show software licenses , it indicates a max of 2 languages are allowed. CUE# sho software licenses Installed license files: - voicemail_lic.sig : 12 MAILBOX LICENSE Core: - Application mode: CCME - Total usable system ports: 6 Voicemail/Auto Attendant: - Max system mailbox capacity time: 840 - 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: 2 - Max enabled languages: 2 ___ 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] Need some help with GK/CUBE Remote Zone call debugging.
The ccapi debug will show you the cause code which doesn't explain why the call failed , you have to debug the h245 asn1 and check the TCS and see the codecs advertised and received and then you will get the TCS negotiation failure so you can explain that there is codec mismatch Ash On Thu, Dec 1, 2011 at 11:55 AM, Mohd Baqari baqari.voic...@gmail.com wrote: Use the command debug voice ccapi inout. H323 debugs won't show in this case. Regards, Mohammed Al Baqari Sent from my iPhone On Dec 1, 2011, at 6:12 PM, ccielabrat ccielab...@gmail.com wrote: I'm trying to setup a call from HQ CUCM via GK-Trunk to a Remote Gk Zone. I have the Gatekeeper configured with OutVia for the remote zone referencing a CUBE on the HQ router. I didn't realize (but it makes sense now) that with Wait for H.245 unchecked on on the CUCM trunk, the call setup goes to the GK/CUBE as g.711. This obviously causes a problem when the CUBE (by default) tries to create the outgoing call leg to the remote zone using G.729. I don't have an XCoder available to CUBE at this point. My problem is that I can't see the codec mismatch failure in debug cch323 h225 or debug cch323 h245. (If it's in there , I'm not seeing :) ) Can someone help me understand if the failure is noted in either of these debugs Or Point me towards a debug that would show the codec mismatch problem? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ 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] Need some help with GK/CUBE Remote Zone call debugging.
20:45:36.810: h245_decode_one_pdu: H245ASNDecodePdu rc = 0, bytesLeftToDecode = 0 Dec 1 20:45:36.810: h245_decode_one_pdu: Read Pkt body: more_pdus:0 rc:0 asn_rc:0 HQ# HQ# HQ# HQ#sho deb H.245: H.245 ASN1 Messages debugging is on On Thu, Dec 1, 2011 at 2:00 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote: The ccapi debug will show you the cause code which doesn't explain why the call failed , you have to debug the h245 asn1 and check the TCS and see the codecs advertised and received and then you will get the TCS negotiation failure so you can explain that there is codec mismatch Ash On Thu, Dec 1, 2011 at 11:55 AM, Mohd Baqari baqari.voic...@gmail.com wrote: Use the command debug voice ccapi inout. H323 debugs won't show in this case. Regards, Mohammed Al Baqari Sent from my iPhone On Dec 1, 2011, at 6:12 PM, ccielabrat ccielab...@gmail.com wrote: I'm trying to setup a call from HQ CUCM via GK-Trunk to a Remote Gk Zone. I have the Gatekeeper configured with OutVia for the remote zone referencing a CUBE on the HQ router. I didn't realize (but it makes sense now) that with Wait for H.245 unchecked on on the CUCM trunk, the call setup goes to the GK/CUBE as g.711. This obviously causes a problem when the CUBE (by default) tries to create the outgoing call leg to the remote zone using G.729. I don't have an XCoder available to CUBE at this point. My problem is that I can't see the codec mismatch failure in debug cch323 h225 or debug cch323 h245. (If it's in there , I'm not seeing :) ) Can someone help me understand if the failure is noted in either of these debugs Or Point me towards a debug that would show the codec mismatch problem? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ 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] Vol 2 Lab 2: cannot get inbound H.323 trunk calls to CME SIP phone to work G.729 to G.711
Hello Anthony , You cannot Transcode call that Hit Dial peer with Voice class codec , it make sense as the router though that he can support Both codecs I hope this clarify the issue you saw Ash On Wed, Nov 30, 2011 at 8:36 PM, Anthony Alba ascanio.al...@gmail.com wrote: Very strange: I can now get both inbound and outbound calls to CME SIP working with transcoder invoked at BR2-RTR. I cannot use voice-class codec 1 under the dial-peer. This surprises me: why would voice class codec hurt the task? voice class codec 1 codec pref 1 g729r8 codec pref 2 g711ulaw If I put this under any of the dial-peers it breaks CME SIP. ___ 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] Live Record
Its not supposed to get involved in this discussion but Edger there was another 100K way to say the same thing in more respectful way Amit Note is Valid and i see Ken accept it and clarify where he come from , i agree this is a open Alias and anyone can ask anything , but please allow me to remind you that the last goal of anyone participate in this alias is to be Expert of Cisco so it will be good to have expert level of question going in here ( any Questions always welcomed) based on my small experience in this Certificate , you have to exhaust all the local resources you have before email the alias because the answer you got in a second will be forgotten in a second later . Respectfully , Ash On Mon, Nov 28, 2011 at 12:20 AM, Edgar Feliz ejzi...@gmail.com wrote: Hi ken, Don't pay attention to A Mitstake... You can ask all the questions you want here.. he is not the boss of us... If he does not want to participate in a discussion he should just keep his trap shut. E On Mon, Nov 28, 2011 at 12:20 AM, Ken Wyan kew...@gmail.com wrote: Dear Sir (Amit), Thanks for your encouragement. Keep it going. Ken ( before asking forum , I saw LiveRcd available in CUCME , but in none of CUCM softkey templates had it or couldn't add for a new softkey template ) On Mon, Nov 28, 2011 at 10:25 AM, Amit Singh batraji...@yahoo.com wrote: Mate I guess u can login into cucm and have a quick look. I am 100% sure that is much faster than typing this email. Mate seems like ur not putting any efforts in studying from your side. Regards Amit Sent from my iPad On 28/11/2011, at 4:52 PM, Ken Wyan kew...@gmail.com wrote: Is Live Record Softkey (LiveRcd) available with CUCM ( Call Manager Server based system ) ? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ 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] Live Record
Agree , if its not yet improved on CCM 8.5 , 8.6 then i am sure that there is enhancement defect opened for this and it will be deployed much better in the newer CCM/CUC releases , i am going through CCM 8.6 these days , if any useful info will show up i will share it here Ash On Mon, Nov 28, 2011 at 2:00 AM, Julien Krieger krieger.jul...@gmail.com wrote: Out of the topic Live record is a nice feature but when you conference in someone with live record, you have these 1/2 secs of MoH (depending on how fast you are) which could be nice to be able to avoid. Could be nice if Cisco had developped a new button/softkey that you could be programmed for multiple CUCM actions at the same time (like conference + speed dial) Julien 2011/11/28 Ashraf Ayyash ash.ayy...@gmail.com Yes Edger , this is correct , its also documented in the CUC Admin guide , Ash On Mon, Nov 28, 2011 at 12:17 AM, Edgar Feliz ejzi...@gmail.com wrote: I think Ash means you have to have a CTI route point with the # you will call to record and have that forward all to VM/CUC. When you are on a call then you can hit the conference soft-key and put in the number mentioned above from the route-point and conference soft-key again to start recording but there is not single key option...not that I am aware off. E PS Amit Questions are allowed to be asked here if you don't want to participate you don't have too mate. On Sun, Nov 27, 2011 at 11:49 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote: NO , you have to create conferance call with the Live record number Ash On Sun, Nov 27, 2011 at 9:52 PM, Ken Wyan kew...@gmail.com wrote: Is Live Record Softkey (LiveRcd) available with CUCM ( Call Manager Server based system ) ? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ 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] CUE intergrated with CUCM in SRST
Hello All, in regard of the CUE MWI will work with Wan outage , this is indeed Supported and it will keep working as the CUE module will fallback to use the SIP Subsystem as alternative protocol and the MWI will work , did you bind the SIP to the correct Interface ( 99% was the reason of this issue ) ? and configure the MWI Server under the sip-ua in the SRST routrer config ? Ash On Mon, Nov 28, 2011 at 10:26 AM, Gurpreet Singh Kukreja tycoononway1...@gmail.com wrote: Hi Raees, Have you created a SIP trigger in CUE as well? You need a JTAPI and SIP trigger in CUE something like this: ccn trigger jtapi phonenumber application voicemail enabled maxsessions 6 end trigger ccn trigger sip phonenumber application voicemail enabled maxsessions 6 end trigger Here, is the Trigger #. Also, have you tried using Outcall in place of unsolicited? If using unsolicited, have you made sure that the SIP UA and mwi sip server commands are present in your config? HTH Regards Gurpreet On Mon, Nov 28, 2011 at 8:28 AM, Raees Shaikh racerra...@yahoo.com wrote: Hi All, When the phones are in the SRST mode, they were able to leave retrieve messages from CUE which was intially integrated with CUCM. However, the MWI dont seem to work. I found that the CUE does not know how to reach the SRST router, as its no where mentioned in the initial or later part of the configuration. Hence I manually entered the gateway under ccn subsystem sip as below ccn subsystem sip gateway address 10.10.202.1 mwi sip unsolicited end subsystem However, the MWI still dont seem to work. On thing to note is that, if I reboot the CUE, everytime it boots up, its sends a SIP Notify for MWI On/Off, but later if I leave a message and refresh the MWI, it does not seem to work. Everything however works perfectly fine when the phones are registered to CUCM. Any pointers? TR ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Live Record
NO , you have to create conferance call with the Live record number Ash On Sun, Nov 27, 2011 at 9:52 PM, Ken Wyan kew...@gmail.com wrote: Is Live Record Softkey (LiveRcd) available with CUCM ( Call Manager Server based system ) ? ___ 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] Live Record
Yes Edger , this is correct , its also documented in the CUC Admin guide , Ash On Mon, Nov 28, 2011 at 12:17 AM, Edgar Feliz ejzi...@gmail.com wrote: I think Ash means you have to have a CTI route point with the # you will call to record and have that forward all to VM/CUC. When you are on a call then you can hit the conference soft-key and put in the number mentioned above from the route-point and conference soft-key again to start recording but there is not single key option...not that I am aware off. E PS Amit Questions are allowed to be asked here if you don't want to participate you don't have too mate. On Sun, Nov 27, 2011 at 11:49 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote: NO , you have to create conferance call with the Live record number Ash On Sun, Nov 27, 2011 at 9:52 PM, Ken Wyan kew...@gmail.com wrote: Is Live Record Softkey (LiveRcd) available with CUCM ( Call Manager Server based system ) ? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CCIE Payment
Go Ahead and contact the Certification Support to track this down , how long has it been since you applied the exam ? I had similar issue when i took the exam and the certification support team sort it out for me Best of Luck Ash On Thu, Nov 24, 2011 at 9:16 PM, Ccie Voice v.c...@yahoo.com wrote: Hi all, I added my credit card to pay for Cisco but they did not proceed the payment. I did not take care about it, because someone told me that Cisco now proceeding the payment after the lab, I went for CCIE lab and I did my lab but till now I did not receive my result and the payment still pending. Any body attempt soon can tell me if he paid before or after the lab? 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 ___ 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] AAR with Route Patterns
This behavior usually mean that you have missing AAR group/CSS somewhere in the call flow you have , also take to the consideration the Bug of the AAR group will only be applied on the Line Level of the phones , Btw the CCM trace will give you a very Nice explanation of why the Call didnt got rerouted . Ash On Thu, Nov 24, 2011 at 9:47 AM, Rrcrumm rrcr...@yahoo.com wrote: Hi Check if the phone devices and have an aar CSS, aar group is on the line . Add both to your gateways and route list can use local route group, make sure aar rp is sending the correct number of digits to PSTN HTH Randall Sent from my iPhone On Nov 24, 2011, at 2:36 AM, Rynard Coetzee rynard.coet...@bytes.co.za wrote: Hi guys I`m trying to get the following scenario to work ,but can`t seem to get the AAR to invoke. I have 2 GW`s out to the PSTN ,they are located at HQ site and BR1 site ,what I have a RPattern that sends all calls from HQ/BR1 phones out of the HQ GW. I then have the same RP`s created in a AAR partition ,pointing out the BR1 GW ,so when I limit my Location bandwidth to 23k so as to not allow the call across the WAN to the remote GW ,it fails the call with “Not Enough Bandwidth” message displaying on the phone but it does not invoke AAR ,so I never see the “Network Congestion ,Rerouting” message and the call just fails. AAR is enabled on the cluster and I can get AAR working between the HQ and BR1 phones so AAR definitely works. Any ideas ? Rynard ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Need help understanding this behavior
I Got real case for the same issue but longtime ago , cannot remember 100% what i have done on it can you please collect ccm sdi sdl and IPVMS detailed traces for the both calls ? I will may take a look when i will have Chance to , but i also need to know why you have enabled the MTP first place and what type of MTP we have IOS or CCM MTP ? Ash On Thu, Nov 24, 2011 at 4:42 PM, Priyank Kiran priyank.ki...@gmail.com wrote: It does On Thursday, November 24, 2011, Mohd Baqari baqari.voic...@gmail.com wrote: The MRGL of MTP should have MoH multicast. Regards, Mohammed Al Baqari Sent from my iPhone On Nov 25, 2011, at 1:57 AM, Priyank Kiran priyank.ki...@gmail.com wrote: No it's not, have 2 MRGLs 1) MRGL attached to DP of gateway and MTP are same mrgl-siteXX mrg-moh + mrg-mtp-siteXX 2) MRGL attached to DP of MoH server mrgl-moh mrg-moh Would like to point out that I only have 1 muticast source and server in my cluster which has been bound to mrg-moh. On Thu, Nov 24, 2011 at 4:44 PM, Mohd Baqari baqari.voic...@gmail.com wrote: What is the MRGL assigned to the device pool of your MTP. Is it the same MoH multicast MRGL. Regards, Mohammed Al Baqari Sent from my iPhone On Nov 24, 2011, at 9:50 PM, Priyank Kiran priyank.ki...@gmail.com wrote: Experts, Need help understanding the following behavior conceptually - Have the subscriber as dedicated MOH multicast server incrementing on port with default address 239.1.1.1 port 16384 Remote H323 gateway, with local music-on-hold wav file spoofing the above source address. This works as expected when put on HOLD and I see all the right output via show ccm-manager music-on-hold and debug ccm-manager music events and show perf query class However, when I check the Media Termination Point Required box on the gateway page in CUCM - I no longer see it sourcing off of the local router flash and it now becomes a unicast stream sourcing off of the Subscriber which I can see from the show perf query class command. Couple questions I have is 1) What forces it to go unicast when you check the MTP required box? 2) Can you still have multicast music-on-hold stream off the local router flash with MTP required check ON? Thanks, Priyank ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CUC VM Mask
Can you give the calling number Translation try ? let me know how it goes Ash On Fri, Nov 25, 2011 at 7:07 AM, datucha123 datucha123 datucha...@gmail.com wrote: As I know, I can use the RDNIS number translation also in SRST Router, so that the correct RDNIS will be passed to CUC. As for Hunt Pilot, it will translate only the Calling number (NOT RDNIS). Sometimes there is a requirement not to use the Alternate Extensions in CUC to acomplish some tasks, so we have either to use the RDNIS translation on the SRST Router, or use the VM Mask Box, if the IP Phones are registered. On Fri, Nov 25, 2011 at 12:38 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote: If i understood correctly , you have CUCM CUC integration and you have SRST Site and you the site went to the SRST and the phone became telephony service phone the VM mask which is configured on the VM profile on the CCM will be no longer effective If this is correct , go ahead and do calling number manipulation on the Hunt Pilot of the VM and this should suffice otherwise you can configure alternative number on the CUC . Ash On Thu, Nov 24, 2011 at 3:29 AM, datucha123 datucha123 datucha...@gmail.com wrote: I have tryed the same with SRST, but that Mask in VM Profile did not take effect :( Can anybody tell me, when does that Mask is activated? For CUCM registered IP Phones it is working file, but for Phones in SRST mode, it does not work any more. On Thu, Nov 24, 2011 at 2:09 AM, Chris Martin clm.c...@gmail.com wrote: Since these are registered with CME and not associated phones with CUCM in any way, I am not sure the voice mail profile is invoked since that is tied to a line. I don't think I have tested CME phones locally registered redirecting into CUCM/CUC without SRST involved. One option would be a voice-translation rule that strips the RDNIS to 4 digits, then associate this to a dial-peer going to voicemail. Chris On Wed, Nov 23, 2011 at 3:14 PM, datucha123 datucha123 datucha...@gmail.com wrote: Yes, sure, it is enabled on the Incoming Gateway. I have also noticed that the Voice Mail Box Mask is working only for the CUCM Registered IP Phones. It transforms the Redirecting and Calling Number as necessary. But when the IP Phone, that is not registered with CUCM (CUCME IP Phones) is redirecting the call to CUC, that Transformation does not work :( On Thu, Nov 24, 2011 at 1:06 AM, Chris Martin clm.c...@gmail.com wrote: At a glance your config seems fine and it should work as you intend. I assume you also allowed incoming redirecting number for the CUCM gateway? Chris On Wed, Nov 23, 2011 at 11:38 AM, datucha123 datucha123 datucha...@gmail.com wrote: Hello, I have configured the Voice Mail Box Mask for (As I know, this will transform the Redricting Number to last 4 digits). but somehow it does not work :( So here what is happening: I have configure the User and VM Box in CUC for CUCME IP Phone, (ext 3012). I have configure the Call Forward All on CUCME IP Phone to CUC VM Pilot through the PSTN (0001911444888) Also configured the Voice Mail Box Mask for VM Porfile to . CUCME IP Phone has also the DialPlan pattern assigned, so that the Caller ID is 2553012. CUCME IP Phone is calling CUC through the PRI PSTN. (I enabled the Redirecting IE on PR interfaces). Also I have enabled the call-forward system redirect in Telephony-service, so that the redirecting number will be expanded based on the DialPlan pattern. So now when the some Phone calls this CUCME IP Phones (That has CFWDALL to CUC PILOT NUMBER Through PSTN) the caller hears the AA Greeting instead of Voice Mail leaving greeting. So as I guess the Voice Mail Box Mask transformation is not working. Maybe I am missing something? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CUC external Transfers
From the restriction table configuration on the CUC to globally enable it , on the CCM make sure you will have CSS of the VM port contain the right PT to make the call hit whatever RP you meant to send to the call to Ash On Fri, Nov 25, 2011 at 5:36 AM, datucha123 datucha123 datucha...@gmail.com wrote: How to enable CUC to transfer to external destinations? By default, when the CUC answers the call we can diall only the Extensions that are presend in CUC users. But how to enable CUC to transfer to any destination? ___ 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] CCIE Payment
its not a rule , it depend on what do you work as in Cisco and on the Center you are working with . Ash On Fri, Nov 25, 2011 at 10:49 AM, Ken Wyan kew...@gmail.com wrote: As I know , Cisco employees could give CCIE Lab Exams free of Exam Cost for 2~3 number of attempts. Is this facility still available? On Fri, Nov 25, 2011 at 1:50 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote: Go Ahead and contact the Certification Support to track this down , how long has it been since you applied the exam ? I had similar issue when i took the exam and the certification support team sort it out for me Best of Luck Ash On Thu, Nov 24, 2011 at 9:16 PM, Ccie Voice v.c...@yahoo.com wrote: Hi all, I added my credit card to pay for Cisco but they did not proceed the payment. I did not take care about it, because someone told me that Cisco now proceeding the payment after the lab, I went for CCIE lab and I did my lab but till now I did not receive my result and the payment still pending. Any body attempt soon can tell me if he paid before or after the lab? 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 ___ 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] I'M DONE!!!
Many congratulation Man , Enjoy the X Box Time :)) Ash CCIE Voice # 31524 On Tue, Nov 22, 2011 at 12:17 PM, Mark Reed marklr...@gmail.com wrote: I forgot the most important part. Mark L Reed CCIE #31990 On Tue, Nov 22, 2011 at 1:16 PM, Mark Reed marklr...@gmail.com wrote: I took my second attempt at RTP yesterday and passed. Thank you to IPExpert for the great study materials. Going through their materials 60 hours a week made this possible. Vik, thank you for the great explanations in the videos. Amy, oh Amy, I'm sorry but I'm really looking forward to hearing someone elses voice coming from my car speakers. Thank you so much to the tips I learned while listening. Special thanks to Matthew Berry. You gave me the inspiration to know it was possible to pass on a first attempt. I almost got it on the first try but did knock it out on the second. Mostly thank you for your videos. I used your Device Based config with just a couple of tweaks to make it my own. Same for the dial plan notes. For those that haven't seen them I would consider them essential viewing. http://www.youtube.com/watch?v=c1OKIJDDcaEfeature=related http://www.youtube.com/watch?feature=player_embeddedv=4mP5powuFUM Now I'm going to go home and open my X-Box 360 that has just been waiting for this day. -- Thanks, Mark L Reed -- Thanks, Mark L Reed Home: 260-637-1585 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] lab - 6
Hello , This is Wrong in the Version you have in the lab , The Software MTP of the CCM can only work on G711 Codec so if you need to have MTP for G729 call , you have to have IOS MTP register to the CCM and available to be allocated for that Call in order for the CCM to use it to do Early Offer , CCM 7.0 is not capable of doing EO on his own , so you have to allocate MTP in the call to make it EO . Starting CCM 8.5 , the CCM can do EO on his OWN as long as you have the latest Version of Phone FW's and IOS Ash On Mon, Nov 21, 2011 at 3:33 PM, Bartosz Sokolowski ibartosz.sokolow...@gmail.com wrote: You are wrong. Early Offer requires MTP. CCM doesn't care how you provide MTP. It's possible to configure software MTP in IOS with G711 or G729. If we consider software MTP it's G711 *or* (not and!) G729. If you use hardware MTP on IOS it's G711 and/or G729. -- Regards, Bartosz 2011/11/21 Peter Jeff peterjeff2...@yahoo.com But early offer required MTP hardware if i am nt wrong *From:* Mohamed Hassan mrmha...@gmail.com *To:* Abel ... midga...@gmail.com *Cc:* ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com *Sent:* Monday, November 21, 2011 12:28 PM *Subject:* Re: [OSL | CCIE_Voice] lab - 6 SIP Early offer it is IOS command, it is equal to the configuration of fast start on CUCM http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb-gw-sipsip.html#wp1351645 It can be configured globally SUMMARY STEPS *1. **enable* *2. **configure terminal* *3. **voice service voip* *4. **allow*-*connections sip* *5. **early-offer forced* *6. **exit* *and it can be configured under dial-peer* SUMMARY STEPS *1. **enable* *2. **configure terminal* *3. **dial-peer voice 1 voip* *4. **voice-class sip early-offer forced* *5. **exit* On Mon, Nov 21, 2011 at 2:48 AM, Abel ... midga...@gmail.com wrote: Well, the DHCP issue can be done easy if you look 5 minutes on cisco.comdocumentation, by Early offer what you mean? no interdigit timeout? On Sun, Nov 20, 2011 at 2:22 PM, datucha123 datucha123 datucha...@gmail.com wrote: Do you mean under the DHCP configuration, that such question has been on the exam? What do you mean under the SIP early offer? or MGCP TS? On Sun, Nov 20, 2011 at 8:24 PM, Peter Jeff peterjeff2...@yahoo.comwrote: Hi Guys, It was my 5th attempt and i got lab 6 frustration is on peak everytime i went for the lab i get new lab From last 2 months all my JR guys passed in dubai bec they got lab 3 and lab 4 in Dubai since last 2 months i saw lot of guys cleared from dubai Now lab 6 i am so so so frustrated its all new Configure the local Cisco router R3 as DHCP server to provide ip add for SC from local subnet , MAke sure SCPH1 get ip add 142.102.66.11 and SCPH2 shld be 142.102.66.12 YOU ARE ALLOWED TO CREATE ONE DHCP POOL FOR ALL SC PHONES SIP early offer , MGCP TS which some different issue. best of luck guys my Christmas will go bad this time as no dates available in Dubai now. Its my bad sunday Regards Jeff ___ 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 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 http://www.platinumplacement.com/ -- Engineer / Mohamed Rabea Unified communication engineer ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] lab - 6
100 percent correct , the lab is based on 7.0 CCM and o have confirmed that the EO support nativly by the CCM is only on 8.5 and further . Ash On Monday, November 21, 2011, William Affeldt william.affe...@yahoo.com wrote: I am 99.9 percent sure that the version in the lab only supports Software MTP at g711. I am sure Vic can confirm. Sent from my iPhone On Nov 21, 2011, at 1:33 PM, Bartosz Sokolowski ibartosz.sokolow...@gmail.com wrote: You are wrong. Early Offer requires MTP. CCM doesn't care how you provide MTP. It's possible to configure software MTP in IOS with G711 or G729. If we consider software MTP it's G711 or (not and!) G729. If you use hardware MTP on IOS it's G711 and/or G729. -- Regards, Bartosz 2011/11/21 Peter Jeff peterjeff2...@yahoo.com But early offer required MTP hardware if i am nt wrong From: Mohamed Hassan mrmha...@gmail.com To: Abel ... midga...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Monday, November 21, 2011 12:28 PM Subject: Re: [OSL | CCIE_Voice] lab - 6 SIP Early offer it is IOS command, it is equal to the configuration of fast start on CUCM http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb-gw-sipsip.html#wp1351645 It can be configured globally SUMMARY STEPS 1. enable 2. configure terminal 3. voice service voip 4. allow-connections sip 5. early-offer forced 6. exit and it can be configured under dial-peer SUMMARY STEPS 1. enable 2. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] SRST Advanced Config
you can use huntstop channel command under the call manager fall back which will limit both in and out number of the calls on the dual/octo lines Ash On Sat, Nov 19, 2011 at 9:24 PM, Ccie Voice v.c...@yahoo.com wrote: Hi all, is it possible to configure: 1- Maximum Number of Calls 2- Busy Trigger In SRST Call-Manger-fallback NOT CME SRST? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] SRST Advanced Config
Hello , you cannot do this with call manager fallback , the hunt stop channel is global command for all the phones and for in/outbound calls on the lines . Ash On Sat, Nov 19, 2011 at 11:13 PM, Ccie Voice v.c...@yahoo.com wrote: Thank you Ashraf for your reply, but could you please help more. if I need to configure the following: Maximum Number of Calls: 4 Busy Trigger 2 how I can configure the above using huntstop channel command? Thanks in advanced. From: Ashraf Ayyash ash.ayy...@gmail.com To: Ccie Voice v.c...@yahoo.com Cc: CCIE Study ccie_voice@onlinestudylist.com Sent: Sunday, November 20, 2011 1:20 AM Subject: Re: [OSL | CCIE_Voice] SRST Advanced Config you can use huntstop channel command under the call manager fall back which will limit both in and out number of the calls on the dual/octo lines Ash On Sat, Nov 19, 2011 at 9:24 PM, Ccie Voice v.c...@yahoo.com wrote: Hi all, is it possible to configure: 1- Maximum Number of Calls 2- Busy Trigger In SRST Call-Manger-fallback NOT CME SRST? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] SRST Advanced Config
you cannot do this in the call manager fallback , you can use use octo line , and hunt stop channel command which will control the incoming calls (busy trigger) and you can have the rest of the octo line for the outgoing call . Ash On Sun, Nov 20, 2011 at 12:34 AM, Ccie Voice v.c...@yahoo.com wrote: thank you for your reply, but I did not understand exactly what I should configure? Could you please explain to me how huntstop channel can solve this? and what I should use huntstop cahnnel 1, 2 .. ? Regards, From: Ashraf Ayyash ash.ayy...@gmail.com To: Ccie Voice v.c...@yahoo.com Cc: CCIE Study ccie_voice@onlinestudylist.com Sent: Sunday, November 20, 2011 2:35 AM Subject: Re: [OSL | CCIE_Voice] SRST Advanced Config Hello , you cannot do this with call manager fallback , the hunt stop channel is global command for all the phones and for in/outbound calls on the lines . Ash On Sat, Nov 19, 2011 at 11:13 PM, Ccie Voice v.c...@yahoo.com wrote: Thank you Ashraf for your reply, but could you please help more. if I need to configure the following: Maximum Number of Calls: 4 Busy Trigger 2 how I can configure the above using huntstop channel command? Thanks in advanced. From: Ashraf Ayyash ash.ayy...@gmail.com To: Ccie Voice v.c...@yahoo.com Cc: CCIE Study ccie_voice@onlinestudylist.com Sent: Sunday, November 20, 2011 1:20 AM Subject: Re: [OSL | CCIE_Voice] SRST Advanced Config you can use huntstop channel command under the call manager fall back which will limit both in and out number of the calls on the dual/octo lines Ash On Sat, Nov 19, 2011 at 9:24 PM, Ccie Voice v.c...@yahoo.com wrote: Hi all, is it possible to configure: 1- Maximum Number of Calls 2- Busy Trigger In SRST Call-Manger-fallback NOT CME SRST? ___ 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] srr-queue bandwidth limit for output priority queue
you can preserve BW using the Shape , but once you enabled the priority Queue out , you will disable the shaping and the sharing and the priority Q will be always served until it will be empty . Ash On Mon, Nov 14, 2011 at 11:34 AM, Ken Wyan kew...@gmail.com wrote: Hi Experts, Is it possible to reserve bandwidth for the priority queue in egress queuing using the command Switch(conf-if)# srr-queue bandwidth shape 4 4 4 4 Switch(conf-if)# priority-queue out In this case will it reserve 25% bandwidth for queue 1 (which is the priority queue) OR will it serve queue 1 until it is empty? As I saw in a cisco doc ( Cisco Catalyst 3750 QoS Configuration Examples ) we can't do bandwidth-limiting of egress queue , unlike ingress queue . Please comment. Ken ___ 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] Question about CUBE Gatekeepers
Hello , you need to Fully understand why you have introduce the CUBE in the Middle between the Zones the CUBE functionality is to be a Via Zone so you can terminate the Signalling/RTP at the CUBE and then re-originate it but with source ip of the CUBE and this is Called Via Zone and so you will need to have allow connection H323 to h323 and in/out dial-peer with session target RAS in this case . Until now there is no need for the telephony service in the cube , when you need Telephone Service ? it will be needed if you have this call setup : CUCM/CME...etc..--incoming dial-peer with g711codecGK/ CUBE(ViaZone)---outbound dial-peer with G729 Codec-CCM/CME/GK.etc.. so now the telephony will be configured to registration a transcoder resource locally configured on the CUBE to translate the call from g711 to G729 and the opposite . Check this DOC for further info in regard of the Via zone concept and the GK in General http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a00800a8928.shtml Ash On Sat, Nov 12, 2011 at 8:25 PM, ccielabrat ccielab...@gmail.com wrote: I need clarification about Gatekeepers using outvia to a CUBE zone. I've always thought a CUBE config needed the underlying Telephony-service config to be operational. Is that the case? I suppose if the call setup is using g.729 in/out of the CUBE , there is no need to have anything but a matching dial-peer and the allow h323 to h323 in the voice service voip. Can someone confirm or correct my understanding. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] can cuc demo license run vpim
i think Cisco Lic Team can help you with that and they are the only people can help , go ahead and contact them and mention that you use the lic on Lab test and they will may issue Temp License with VPIM enabled Ash 2011/11/9 bruno bruno.juni...@gmail.com: how can i get it for labbing . i do it in my home lab -- Best Regards, Bruno -- Original -- From: Farkas Péterwormh...@sch.bme.hu; Date: Wed, Nov 9, 2011 07:50 PM To: brunobruno.juni...@gmail.com; Cc: CCIE-V邮件列表ccie_voice@onlinestudylist.com; Subject: Re: [OSL | CCIE_Voice] can cuc demo license run vpim No, demo license not cover VPIM so it requries VPIM license to be added. However proctorlabs should have. Peter - Original Message - From: bruno bruno.juni...@gmail.com Date: Wednesday, November 9, 2011 11:18 am Subject: [OSL | CCIE_Voice] can cuc demo license run vpim To: CCIE-V邮件列表 ccie_voice@onlinestudylist.com When I attempt to add a VPIM location is Unity Connection I receive the following license error. Anyone attempt VPIM in these labs yet? Status The requested operation would result in a license violation. Unable to create VPIM Location -- Best Regards, Bruno ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Downloading List.xml from Call Manager.
Exactly Chris , Basant , the right format is List.xml not list.xml , William , if we have the TFTP traces everything will be very clear Ash On Mon, Oct 24, 2011 at 3:51 PM, Chris Martin clm.c...@gmail.com wrote: Just to add, remember everything is case sensitive.. I have shot myself in the foot more than once when uploading files, both the name and the directory structure. IE: Desktops/320x212x12 does not equal desktops/320x312x12.. If placed in the wrong spot the phone will not find it. You can see what is being requested in UCM traces. Chris On Mon, Oct 24, 2011 at 12:32 PM, William Affeldt william.affe...@yahoo.com wrote: Ok, so I have to be missing something. I can't download the List.xml file from Call Manager but I can download my image files. I have triple checked the file name and the directory. I am downloading the file from same directory as the images and they work fine. Is there something I am missing to download the file? Also, I downloaded different .xml file just fine. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Downloading List.xml from Call Manager.
unfortunately i dont have lab right now that's why my answers is weak because i relay on my memory when i studied this feature , but why you are uploading the List.xml to the root ? it should place on the same directory as the Photos Ash On Mon, Oct 24, 2011 at 4:12 PM, William Affeldt william.affe...@yahoo.com wrote: Hi All, Thank you for the replys. The error from RTMT is File[List.xml] not found. Also, all of you are referring to sending the file to the phone. I am trying to download the file from call manager to BR2. I know the error code seems strait forward but it is not. The file List.xml is in the root directory. I have also uploaded three other files into the same / directory and they work fine. I have deleted it and uploaded it again. I have had two other sets of eyes look at the spelling and directory placement. Can someone else try to upload the List.xml file into the root directory and download it to BR2. From: Chris Martin clm.c...@gmail.com To: William Affeldt william.affe...@yahoo.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Monday, October 24, 2011 1:51 PM Subject: Re: [OSL | CCIE_Voice] Downloading List.xml from Call Manager. Just to add, remember everything is case sensitive.. I have shot myself in the foot more than once when uploading files, both the name and the directory structure. IE: Desktops/320x212x12 does not equal desktops/320x312x12.. If placed in the wrong spot the phone will not find it. You can see what is being requested in UCM traces. Chris On Mon, Oct 24, 2011 at 12:32 PM, William Affeldt william.affe...@yahoo.com wrote: Ok, so I have to be missing something. I can't download the List.xml file from Call Manager but I can download my image files. I have triple checked the file name and the directory. I am downloading the file from same directory as the images and they work fine. Is there something I am missing to download the file? Also, I downloaded different .xml file just fine. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] cue-cme mwi unsolicited question
Glad its all sorted out now , Thanks Ash On Sun, Oct 23, 2011 at 8:57 PM, ccielabrat ccielab...@gmail.com wrote: Hey Ashraf, You got me thinking the right way. I had a mismatch between my sip interface and the gateway configured on CUE. Thanks! On Sat, Oct 22, 2011 at 4:43 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote: did you binded the SIP to the correct interface from the CME config Voice service Voip ? Any chance to reload the Funky CUE ? Ash On Sat, Oct 22, 2011 at 12:12 PM, ccielabrat ccielab...@gmail.com wrote: I can't get CUE MWI working either. This is my cue config for SIP ccn subsystem sip gateway address 10.1.131.1 mwi envelope-info mwi sip unsolicited end subsystem I've tried all kinds of config on the CME router without success. When running debug ccsip messages on the CME router , I don't see anything if I issue mwi refresh all on CUE, even though I can dial into CUE and check to hear a voicemail on dn 4001 On Fri, Oct 21, 2011 at 6:47 PM, Brian btmulg...@gmail.com wrote: hi - this is an excellent summary of mwi for cue that is worth a read http://blog.ipexpert.com/2010/07/19/sip-mwi-mechansims-on-cue-notify/ Sent from my iPad On 21 Oct 2011, at 21:23, Ashraf Ayyash ash.ayy...@gmail.com wrote: Hello Zamuel , the mwi relay command is only needed in case of the subscribe notify MWI and its not needed in case of using Unsolicited because it does send the event to the phone using NOTIFY message no matter it subscribed to the MWI server Or not . Ash On Fri, Oct 21, 2011 at 11:21 AM, zamuel del Toro sdelto...@hotmail.com wrote: Hi Vic, how is it going?, about mwi unsolicited. sip-ua mwi.. unsolicited telephony-ser mwi relay ephone-dn nothing works mwi if subscribe notify sip-ua mwi... telephony-ser nothing ephone-dn mwi sip both works fine what if make mistake if on unsolicited include on ephone-dn mwi sip, that work too. is wrong do this? thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ 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] cue-cme mwi unsolicited question
did you binded the SIP to the correct interface from the CME config Voice service Voip ? Any chance to reload the Funky CUE ? Ash On Sat, Oct 22, 2011 at 12:12 PM, ccielabrat ccielab...@gmail.com wrote: I can't get CUE MWI working either. This is my cue config for SIP ccn subsystem sip gateway address 10.1.131.1 mwi envelope-info mwi sip unsolicited end subsystem I've tried all kinds of config on the CME router without success. When running debug ccsip messages on the CME router , I don't see anything if I issue mwi refresh all on CUE, even though I can dial into CUE and check to hear a voicemail on dn 4001 On Fri, Oct 21, 2011 at 6:47 PM, Brian btmulg...@gmail.com wrote: hi - this is an excellent summary of mwi for cue that is worth a read http://blog.ipexpert.com/2010/07/19/sip-mwi-mechansims-on-cue-notify/ Sent from my iPad On 21 Oct 2011, at 21:23, Ashraf Ayyash ash.ayy...@gmail.com wrote: Hello Zamuel , the mwi relay command is only needed in case of the subscribe notify MWI and its not needed in case of using Unsolicited because it does send the event to the phone using NOTIFY message no matter it subscribed to the MWI server Or not . Ash On Fri, Oct 21, 2011 at 11:21 AM, zamuel del Toro sdelto...@hotmail.com wrote: Hi Vic, how is it going?, about mwi unsolicited. sip-ua mwi.. unsolicited telephony-ser mwi relay ephone-dn nothing works mwi if subscribe notify sip-ua mwi... telephony-ser nothing ephone-dn mwi sip both works fine what if make mistake if on unsolicited include on ephone-dn mwi sip, that work too. is wrong do this? thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ 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] cue-cme mwi unsolicited question
Hello Zamuel , the mwi relay command is only needed in case of the subscribe notify MWI and its not needed in case of using Unsolicited because it does send the event to the phone using NOTIFY message no matter it subscribed to the MWI server Or not . Ash On Fri, Oct 21, 2011 at 11:21 AM, zamuel del Toro sdelto...@hotmail.com wrote: Hi Vic, how is it going?, about mwi unsolicited. sip-ua mwi.. unsolicited telephony-ser mwi relay ephone-dn nothing works mwi if subscribe notify sip-ua mwi... telephony-ser nothing ephone-dn mwi sip both works fine what if make mistake if on unsolicited include on ephone-dn mwi sip, that work too. is wrong do this? thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CUPC does not show status of user logged in via IPPM
Hello ALL , Mohammad , this is incorrect , you got to either specify it from the ccm service parameter or from the CUPS server and this is mandatory , Raees , lets get a Quick webex session to check this whenever you have time Ash On Thu, Oct 20, 2011 at 2:09 AM, Mohammed Al baqari baqari.voic...@gmail.com wrote: Hi Pithog , The service parameter in CUCM will be automatically updated once you configure the SIP trunk in Presence. Regards, Mohd Baqari On Thu, Oct 20, 2011 at 4:00 AM, pithog...@yahoo.com wrote: Hi Ash, If you have more than one sip trunk on your call manager, you will have to specify which of them you are using for your presence in enterprise parameters . I hope this helps Pithog oil Sent from my BlackBerry® Smartphone, from Etisalat. Enjoy high speed internet service with Etisalat easy net, available at all our experience centres ___ 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] SIP Trunk with CUC and possible AAR
Hey Inder , Nice catch on this , the AAR is between 2 endpoint register to the same ccm so you cannot use the same concept of redundancy in this config scenario , you can achieve this by having 2 route in the route group , so the sip trunk to the CUC and also GW (of course manipulate the calling and the called to be fit to the PSTN world) and change the service parameter of stop routing when no bandwidth to false , and so you will accomplish the redundancy in this config scenario Ash On Thu, Oct 20, 2011 at 10:48 AM, Inder Singh ising...@gmail.com wrote: Hello All, I am working on a lab that requires to set up CUCM and CUC using SIP Trunk. It then asks for calls that are sent to VM from BR1 to be redirected out the PSTN when there is WAN congestion. I have looked high and low but I can't find any reference where this can be done with AAR...or am I totally missing something. If AAR is possible can someone point me in the right direction? If it is not possible can someone let me know how you might achieve this otherwise? I tried using a route list with the SIP trunk as the primary RG and the PSTN GW and the secondary RG. The issue is redirecting the caller, called and redirect on no answer. We need the BR1 phone to be able to press the message key and retrieve messages (this I was able to do with alternate extensions) and also for callers to be redirected to voicemail for the called party (this I was not able to do with the route list scenario). Thanks in advance for any help you can provide. Regards. Inder. ___ 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] SIP Trunk with CUC and possible AAR
i guess you can take of the redirected number from the call and then do mask on the VM pilot /profile or interduce tranlation pattern in between to match the called and change the calling to 4 digits Ash On Thu, Oct 20, 2011 at 3:57 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote: Hey Inder , Nice catch on this , the AAR is between 2 endpoint register to the same ccm so you cannot use the same concept of redundancy in this config scenario , you can achieve this by having 2 route in the route group , so the sip trunk to the CUC and also GW (of course manipulate the calling and the called to be fit to the PSTN world) and change the service parameter of stop routing when no bandwidth to false , and so you will accomplish the redundancy in this config scenario Ash On Thu, Oct 20, 2011 at 10:48 AM, Inder Singh ising...@gmail.com wrote: Hello All, I am working on a lab that requires to set up CUCM and CUC using SIP Trunk. It then asks for calls that are sent to VM from BR1 to be redirected out the PSTN when there is WAN congestion. I have looked high and low but I can't find any reference where this can be done with AAR...or am I totally missing something. If AAR is possible can someone point me in the right direction? If it is not possible can someone let me know how you might achieve this otherwise? I tried using a route list with the SIP trunk as the primary RG and the PSTN GW and the secondary RG. The issue is redirecting the caller, called and redirect on no answer. We need the BR1 phone to be able to press the message key and retrieve messages (this I was able to do with alternate extensions) and also for callers to be redirected to voicemail for the called party (this I was not able to do with the route list scenario). Thanks in advance for any help you can provide. Regards. Inder. ___ 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] CUPC does not show status of user logged in via IPPM
can you make sure you have associated the LINE of that phone with the User ? Ash On Wed, Oct 19, 2011 at 3:53 AM, Raees Shaikh racerra...@yahoo.com wrote: i tried but still the same. Also, if I change the status of the CUPS, it shows up on the phone instantly, but the phone always shows as offline on the CUPC Regards, Raees Shaikh From: Adil Shaikh adil.sha...@gmail.com To: Raees Shaikh racerra...@yahoo.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Wednesday, October 19, 2011 2:17 PM Subject: Re: [OSL | CCIE_Voice] CUPC does not show status of user logged in via IPPM if everything configure correctly the reset the sip trunk to cups. -adil On Wed, Oct 19, 2011 at 4:46 PM, Raees Shaikh racerra...@yahoo.com wrote: Hi All, I am practicing CUPS after going through the video, I configured IPPM as described in the Video. Everything works fine except the point wherein the CUPC is unable to show the status of user logged in Via IPPM even though I have set the status on the phone to available have this user added on my CUPC. Any help will be appreciated. Thanks Regards, Raees Shaikh ___ 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 -- . . . . _7___|___|_|_|adil.sha...@gmail.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] CUE user password
username XX pin XXX Ash On Tue, Oct 18, 2011 at 6:20 AM, darshan ccievoice0...@hotmail.com wrote: Dear ; Just like creating user and assigning phone number in CUE.. Should we create the password for SC PH1 and SC PH2.if it didn’t ask to assign the password username SCPH2 create username SCPH1 create username SCPH2 phonenumber 4002 username SCPH1 phonenumber 4001 Regards Darsh ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] SRST Behaviour
This can happen if you don't have enough DSP's in the router , can you try to do SRST in max dn 1 and max phone 1 ? Btw the fw of the phone cannot be the issue , the Ios us the had boy here :) Ash On Monday, October 17, 2011, mgscip gpsvoiceexpe...@yahoo.com wrote: Hi , I have some issue in SRST . When the Phones are get into SRST fallback-mode Phones didn't get any DN. I given the SRST mode auto-provision all , but i couldn't see any Ephone configuration in the running configuration. tried with Firmware upgrade , Reload the router but no luck. Thanks, Sriram.P ___ 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] Can someone tell me how to get a evual license for gatekeeper. the license expired
if you have valid CCO ID you may contact Cisco LIC Team and ask them , they are the only people can decide /give you the LIC Ash On Mon, Oct 17, 2011 at 3:10 PM, Cecil Wilson cecil...@gmail.com wrote: Hi Can someone tell me how to get a license for gatekeeper that will not expire. I just need this for lab purposes ___ 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] MOH multicast
hello All , this is called MMOH Snooping and the ccm is a victim , he have no idea about what you do in the Multicast moh stream , so the answer here is not fully accurate , but you will see the counter increament in the show perf query command because the ccm think he is streaming the moh and he is but we are stopping the stream from reaching the phone . the fact is you are configueding the router to stream MOH from the flash of the router and you are using the same exact ip and port of what the ccm using to stream the MOH , so the Phone will request to have MOH stearmed and you will see the ccm respoding with skinny message to listen to the ip and the port you have configued on the ccm so the phone will try to subscibe to ths stream which is already played from the flash and so you will get the mmoh from the flash , at this point of time , the ccm think he is streaming the moh from himself and thats why we calling this snooping . you can do deb ephone moh shwich will show you the stream and the ip add that you flood the music to by command the multicast and you can collect sniffer trace from the phone to see the multicast traffic as well . Ash On Fri, Oct 14, 2011 at 2:50 PM, Mohammed Al baqari baqari.voic...@gmail.com wrote: You are requesting MoH server from CUCM so the count will be increased in CUCM. However, you are hearing the stream from BR1 flash . This is continuously played and once the channel is opened on phone by CUCM to get the multicast stream you will hear the multicast coming from flash On Fri, Oct 14, 2011 at 11:44 PM, Ccie Voice v.c...@yahoo.com wrote: But I am using router flash memory Not CUCM -- *From: *Mohammed Al baqari baqari.voic...@gmail.com; *To: *Ccie Voice v.c...@yahoo.com; *Cc: *CCIE Study ccie_voice@onlinestudylist.com; *Subject: *Re: [OSL | CCIE_Voice] MOH multicast *Sent: *Fri, Oct 14, 2011 3:24:18 PM use the CLI command [show perf query class Cisco MOH Device] on CUCM CLI interface. The count as below should be increased. This will reflect in CUCM and not on BR1 router. admin:show perf query class Cisco MOH Device ==query class : - Perf class (Cisco MOH Device) has instances and values: MOH_2 - MOHHighestActiveResources = 1 MOH_2 - MOHMulticastResourceActive = 1 MOH_2 - MOHMulticastResourceAvailable = 25 MOH_2 - MOHOutOfResources = 0 MOH_2 - MOHTotalMulticastResources = 25 MOH_2 - MOHTotalUnicastResources = 250 MOH_2 - MOHUnicastResourceActive = 0 MOH_2 - MOHUnicastResourceAvailable= 250 On Fri, Oct 14, 2011 at 10:14 AM, Ccie Voice v.c...@yahoo.com wrote: Hi everybody, I am trying to configure Multicast MOH in Branch 1 router, I added the file to router flash and I configure CUCM to use multicast. when I am calling from one phone to another in BR1 I am able to hear the file that I uploaded to router flash memory. but if I tried to use the following command: sho ccm-manager music-on-hold the result: Current active multicast sessions : 0 is it using Multicast? how I can make sure that I am using multicast not unicast? ___ 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 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
Re: [OSL | CCIE_Voice] IP Blue (running multiple instances)
google doesnt have any answer for this ? Ash On Fri, Oct 14, 2011 at 1:10 PM, Jeferson Guardia jefers...@gmail.com wrote: Hi, I am able to register many phones, do the regedit thing etc. but the only thing that doesnt work for me is the /d added into the path at the end of the shortcut command. It appears that VTGO PC Lite doesnt work loading up many instances for IP Blue on Windows 7. Any ideas? -- Jeferson Guardia CCIE #28157 ___ 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] Low score on certain sections.
and even though i got 100% for alot of things i didnt pass and got low scores fer stuff like VM, High availibilty. Now i tested all these and they worked as asked. I was wondering if anyone can give me any ideas on what could have caused such low scores. Thanks Kev -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: 11 October 2011 05:22 To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 68, Issue 71 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: Fwd: Fractional MGCP (Ashraf Ayyash) -- Message: 1 Date: Mon, 10 Oct 2011 23:22:25 -0500 From: Ashraf Ayyash ash.ayy...@gmail.com To: Kshitij Singhi martinian.ksin...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Fwd: Fractional MGCP Message-ID: CAEW==nt4-GYD-spnRN_FOD+==mailto:s9kdvfpplsibyvtlkpy%2b%2b6...@mail.gmail.com Content-Type: text/plain; charset=ISO-8859-1 sorry small correction On Mon, Oct 10, 2011 at 11:19 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote: Hey Kshitij , the service parameter WILL NOT come to play in the SRST as you are taking the ccm out of the game and bringing the default routing application in charge (h323) , the service parameter will come to play when you will have for example 3 bchannal group configured under the controller and you have no service parameter on the ccm ( so the ccm thought he have full T1/E1) ?, and then get 3 concurrent calls (so full of what you have) and then try to setup the 4th call , the ccm will send the call over (please put one GW in the RG) and then you will see Setup going on Channel 4 and you will see the call getting disconnected with requested channel not available and you MUST have configured 3 channels only configured on the PSTN router (which will emulate the real life as the PSTN will setup only channels # of what you paid for and if you will sent new call ( we have 3 active calls) on the 4rth Channel you will get i mentioned about ) and hence the need of the service parameter . finally , in your example i don't think the GW debugs is the good place to check as the route Group is A CCM decision and the GW is MGCP Slave to the CCM so please include ccm sdi /SDL traces on the detailed level so we can see how the ccm decided to send the call to specific endpoint in the route group Ash On Mon, Oct 10, 2011 at 4:59 PM, Kshitij Singhi martinian.ksin...@gmail.com wrote: Resending since the email bounced due to its size. I guess Ken would have received the endpoint configuration and hence removing the attachments. -- Forwarded message -- From: Kshitij Singhi martinian.ksin...@gmail.com Date: Mon, Oct 10, 2011 at 7:36 PM Subject: Re: [OSL | CCIE_Voice] Fractional MGCP To: Ken Wyan kew...@gmail.com Cc: ccie_voice@onlinestudylist.com Attached are the screenshots of the GW configuration (main page and endpoint). Site A and Site B are more or less identical (except for the domain names) and both have the defaults configured (SF set to 4 and Display IE delivery, Redirecting IE delivery outbound/inbound being checked). I performed the following tests: 1. Maxing out the channels and then checking if it fails over to the next option in the RG. 2. Maxing out the channels and then checking if it fails over to the next option in the RL. 3. Maxing out the channels by making incoming calls and then calling out to check if the call goes out via the next option in the RG. 4. Calls during SRST. (both incoming and outgoing to see if there is any change in behavior) 5. Bringing the Site out of SRST to see if there is any change in behavior. FOR POINT 1 GIVEN ABOVE Call 1 going out via channel 3 == Oct 10 12:57:30.789: MGCP Packet received from 192.168.10.47:2427--- CRCX 463 S2/DS1-0/3...@site-b.yourdomain.com MGCP 0.1 C: D202e65000F50003 X: 3 L: p:20, a:PCMU, s:off, t:b8, fxr/fx:t38 M: recvonly R: D/[0-9ABCD*#] Q: process,loop --- Oct 10 12:57:30.813: ISDN Se2/0:23 Q931: TX - SETUP pd = 8 ?callref = 0x0003 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i
Re: [OSL | CCIE_Voice] Cisco MVA
what css you have assigned to the Destination profile ? the css will be used for MVA and reroute css will be used for SNR , please check this, if all is fine , collect detailed level ccm sdi/sdl traces and send it over with the call info , i will may take a look Ash On Tue, Oct 11, 2011 at 9:54 PM, Cisco Nut rafayc...@gmail.com wrote: Hello I have configured Cisco Mobile Voice Access as per V1 Lab 5C, when I dial 2123945999 I hear prompt and after authentication, I enter 1002# I hear a message your call can not be completed as dialed, even if I dial 5001 I get same message. 5999 belongs to internal pt, which in turn member of appropriate CSS. ___ 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] Cisco MVA
can you please make sure you have the DP of the RDP contain the ccm group of both server the pub and the sub ? i will look further in the traces Ash On Wed, Oct 12, 2011 at 8:22 PM, Cisco Nut rafayc...@gmail.com wrote: Hi I call from 2123942123 to 2123945999, it prompts me for my PIN ie 12345#, after pressing 1, I tried calling 5001#, 1001#, 1002# On Wed, Oct 12, 2011 at 9:20 PM, Cisco Nut rafayc...@gmail.com wrote: Hi Ash I have checked CSS on RDP, it is setup as International CSS, which contains Part. for HQ phones and Br1 Phones as well as RP to call Local, LD, and International. Please find the attach trace files. On Wed, Oct 12, 2011 at 1:59 AM, Ashraf Ayyash ash.ayy...@gmail.com wrote: what css you have assigned to the Destination profile ? the css will be used for MVA and reroute css will be used for SNR , please check this, if all is fine , collect detailed level ccm sdi/sdl traces and send it over with the call info , i will may take a look Ash On Tue, Oct 11, 2011 at 9:54 PM, Cisco Nut rafayc...@gmail.com wrote: Hello I have configured Cisco Mobile Voice Access as per V1 Lab 5C, when I dial 2123945999 I hear prompt and after authentication, I enter 1002# I hear a message your call can not be completed as dialed, even if I dial 5001 I get same message. 5999 belongs to internal pt, which in turn member of appropriate CSS. ___ 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] Fwd: Fractional MGCP
14:17:40.908: ISDN Se2/0:23 Q931: TX - CONNECT_ACK pd = 8 callref = 0x0 003 Site-B(config)# Site-B(config)# Site-B(config)# Site-B(config)# Site-B(config)# Site-B(config)#3 calls active ^ % Invalid input detected at '^' marker. Site-B(config)# Site-B(config)# Site-B(config)# Oct 11 14:18:04.074: ISDN Se2/0:23 Q931: TX - DISCONNECT pd = 8 callref = 0x00 03 Cause i = 0x8290 - Normal call clearing Oct 11 14:18:04.082: ISDN Se2/0:23 Q931: RX - RELEASE pd = 8 callref = 0x8003 Oct 11 14:18:04.118: ISDN Se2/0:23 Q931: TX - RELEASE_COMP pd = 8 callref = 0x 0003 Oct 11 14:18:07.894: ISDN Se2/0:23 Q931: TX - DISCONNECT pd = 8 callref = 0x00 01 Cause i = 0x8290 - Normal call clearing Oct 11 14:18:07.902: ISDN Se2/0:23 Q931: RX - RELEASE pd = 8 callref = 0x8001 Oct 11 14:18:07.938: ISDN Se2/0:23 Q931: TX - RELEASE_COMP pd = 8 callref = 0x 0001 Oct 11 14:18:09.102: ISDN Se2/0:23 Q931: TX - DISCONNECT pd = 8 callref = 0x00 02 Cause i = 0x8290 - Normal call clearing Oct 11 14:18:09.106: ISDN Se2/0:23 Q931: RX - RELEASE pd = 8 callref = 0x8002 Oct 11 14:18:09.158: ISDN Se2/0:23 Q931: TX - RELEASE_COMP pd = 8 callref = 0x 0002 After 3 calls were active (and I had only one GW in the RG), I tried a fourth call. I DID NOT see the setup going out of the GW on the 4th call, as suggested by you (and DID NOT get the consequent error requested circuit/channel not available). The call failed with High traffic, please try again later on the IP Phone. I don't see the call hitting the GW at all (not in MGCP nor in the ISDN debugs). Clearly, the CUCM is NOT forwarding the call to the GW (why would it? since the show perf query class clearly shows that the other channels have been marked as unknown). finally , in your example i don't think the GW debugs is the good place to check as the route Group is A CCM decision and the GW is MGCP Slave to the CCM so please include ccm sdi /SDL traces on the detailed level so we can see how the ccm decided to send the call to specific endpoint in the route group The GW debugs was just to illustrate the point that irrespective of HOW is CUCM selecting the backup GW, it definitely is selecting it. The whole point that calls fail if the Service Parameter is not configured in the backup scenario seems to be a myth. I'm pretty sure that the proctors won't be taking the CUCM traces to figure that out (I specifically asked the proctor how do they check the lab when it comes to specific requirements, and he stated that it was mainly manual testing i.e. making/receiving calls and in the case of checking plan/type/digits sent across they usually take a Q.931 debug). As we are well aware, if calls are sent with the incorrect plan/type it fails in the actual lab. Also, as you have correctly stated in the past, L3 on the ISDN leg is CUCM controlled so whatever we see in Q.931 is a reflection of CUCM's decision to route the call. Attaching the SDI/SDL traces will not be possible since there is a cap on the size of the emails that we can send to the OSL and it will be counterproductive since the GW debugs are VERY clear with respect to CUCMs decision to select a particular endpoint. On Tue, Oct 11, 2011 at 9:49 AM, Ashraf Ayyash ash.ayy...@gmail.com wrote: Hey Kshitij , the service parameter come to play in the SRST as you are taking the ccm out of the game and bringing the default routing application in charge (h323) , the service parameter will come to play when you will have for example 3 bchannal group configured under the controller and you have no service parameter on the ccm ( so the ccm thought he have full T1/E1) , and then get 3 concurrent calls (so full of what you have) and then try to setup the 4th call , the ccm will send the call over (please put one GW in the RG) and then you will see Setup going on Channel 4 and you will see the call getting disconnected with requested channel not available and you MUST have configured 3 channels only configured on the PSTN router (which will emulate the real life as the PSTN will setup only channels # of what you paid for and if you will sent new call ( we have 3 active calls) on the 4rth Channel you will get i mentioned about ) and hence the need of the service parameter . finally , in your example i don't think the GW debugs is the good place to check as the route Group is A CCM decision and the GW is MGCP Slave to the CCM so please include ccm sdi /SDL traces on the detailed level so we can see how the ccm decided to send the call to specific endpoint in the route group Ash On Mon, Oct 10, 2011 at 4:59 PM, Kshitij Singhi martinian.ksin...@gmail.com wrote: Resending since the email bounced due to its size. I guess Ken would have received the endpoint configuration and hence removing the attachments. -- Forwarded message -- From: Kshitij Singhi martinian.ksin...@gmail.com Date: Mon
Re: [OSL | CCIE_Voice] [UCCX COMPONENT ACTIVATION]
ALL On Mon, Oct 10, 2011 at 12:43 AM, michael.se...@compucom.com wrote: Where can I check in UCCX to determine what components are activated? Thanks --ms ___ 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] Fwd: Fractional MGCP
Hey Kshitij , the service parameter come to play in the SRST as you are taking the ccm out of the game and bringing the default routing application in charge (h323) , the service parameter will come to play when you will have for example 3 bchannal group configured under the controller and you have no service parameter on the ccm ( so the ccm thought he have full T1/E1) , and then get 3 concurrent calls (so full of what you have) and then try to setup the 4th call , the ccm will send the call over (please put one GW in the RG) and then you will see Setup going on Channel 4 and you will see the call getting disconnected with requested channel not available and you MUST have configured 3 channels only configured on the PSTN router (which will emulate the real life as the PSTN will setup only channels # of what you paid for and if you will sent new call ( we have 3 active calls) on the 4rth Channel you will get i mentioned about ) and hence the need of the service parameter . finally , in your example i don't think the GW debugs is the good place to check as the route Group is A CCM decision and the GW is MGCP Slave to the CCM so please include ccm sdi /SDL traces on the detailed level so we can see how the ccm decided to send the call to specific endpoint in the route group Ash On Mon, Oct 10, 2011 at 4:59 PM, Kshitij Singhi martinian.ksin...@gmail.com wrote: Resending since the email bounced due to its size. I guess Ken would have received the endpoint configuration and hence removing the attachments. -- Forwarded message -- From: Kshitij Singhi martinian.ksin...@gmail.com Date: Mon, Oct 10, 2011 at 7:36 PM Subject: Re: [OSL | CCIE_Voice] Fractional MGCP To: Ken Wyan kew...@gmail.com Cc: ccie_voice@onlinestudylist.com Attached are the screenshots of the GW configuration (main page and endpoint). Site A and Site B are more or less identical (except for the domain names) and both have the defaults configured (SF set to 4 and Display IE delivery, Redirecting IE delivery outbound/inbound being checked). I performed the following tests: 1. Maxing out the channels and then checking if it fails over to the next option in the RG. 2. Maxing out the channels and then checking if it fails over to the next option in the RL. 3. Maxing out the channels by making incoming calls and then calling out to check if the call goes out via the next option in the RG. 4. Calls during SRST. (both incoming and outgoing to see if there is any change in behavior) 5. Bringing the Site out of SRST to see if there is any change in behavior. FOR POINT 1 GIVEN ABOVE Call 1 going out via channel 3 == Oct 10 12:57:30.789: MGCP Packet received from 192.168.10.47:2427--- CRCX 463 S2/DS1-0/3...@site-b.yourdomain.com MGCP 0.1 C: D202e65000F50003 X: 3 L: p:20, a:PCMU, s:off, t:b8, fxr/fx:t38 M: recvonly R: D/[0-9ABCD*#] Q: process,loop --- Oct 10 12:57:30.813: ISDN Se2/0:23 Q931: TX - SETUP pd = 8 callref = 0x0003 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98383 Exclusive, Channel 3 Calling Party Number i = 0x0081, '3002' Plan:Unknown, Type:Unknown Called Party Number i = 0x80, '911' Plan:Unknown, Type:Unknown Oct 10 12:57:30.829: ISDN Se2/0:23 Q931: RX - CALL_PROC pd = 8 callref = 0x8003 Channel ID i = 0xA98383 Exclusive, Channel 3 Call 2 going out via channel 2 === Oct 10 12:57:39.806: MGCP Packet received from 192.168.10.47:2427--- CRCX 467 S2/DS1-0/2...@site-b.yourdomain.com MGCP 0.1 C: D202e65300F50004 X: 2 L: p:20, a:PCMU, s:off, t:b8, fxr/fx:t38 M: recvonly R: D/[0-9ABCD*#] Q: process,loop --- Oct 10 12:57:39.826: ISDN Se2/0:23 Q931: TX - SETUP pd = 8 callref = 0x0004 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98382 Exclusive, Channel 2 Calling Party Number i = 0x0081, '3002' Plan:Unknown, Type:Unknown Called Party Number i = 0x80, '911' Plan:Unknown, Type:Unknown Oct 10 12:57:39.842: ISDN Se2/0:23 Q931: RX - CALL_PROC pd = 8 callref = 0x8004 Channel ID i = 0xA98382 Exclusive, Channel 2 Call 3 going out via channel 1 == Oct 10 12:57:49.279: MGCP Packet received from 192.168.10.47:2427--- CRCX 471 S2/DS1-0/1...@site-b.yourdomain.com MGCP 0.1 C: D202e65600F50005 X: 1 L: p:20, a:PCMU, s:off, t:b8, fxr/fx:t38 M: recvonly R: D/[0-9ABCD*#] Q: process,loop --- Oct 10 12:57:49.299: ISDN Se2/0:23 Q931: TX - SETUP pd = 8 callref = 0x0005 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Calling Party Number i = 0x0081, '3002' Plan:Unknown, Type:Unknown Called Party Number i = 0x80,
Re: [OSL | CCIE_Voice] Fwd: Fractional MGCP
sorry small correction On Mon, Oct 10, 2011 at 11:19 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote: Hey Kshitij , the service parameter WILL NOT come to play in the SRST as you are taking the ccm out of the game and bringing the default routing application in charge (h323) , the service parameter will come to play when you will have for example 3 bchannal group configured under the controller and you have no service parameter on the ccm ( so the ccm thought he have full T1/E1) , and then get 3 concurrent calls (so full of what you have) and then try to setup the 4th call , the ccm will send the call over (please put one GW in the RG) and then you will see Setup going on Channel 4 and you will see the call getting disconnected with requested channel not available and you MUST have configured 3 channels only configured on the PSTN router (which will emulate the real life as the PSTN will setup only channels # of what you paid for and if you will sent new call ( we have 3 active calls) on the 4rth Channel you will get i mentioned about ) and hence the need of the service parameter . finally , in your example i don't think the GW debugs is the good place to check as the route Group is A CCM decision and the GW is MGCP Slave to the CCM so please include ccm sdi /SDL traces on the detailed level so we can see how the ccm decided to send the call to specific endpoint in the route group Ash On Mon, Oct 10, 2011 at 4:59 PM, Kshitij Singhi martinian.ksin...@gmail.com wrote: Resending since the email bounced due to its size. I guess Ken would have received the endpoint configuration and hence removing the attachments. -- Forwarded message -- From: Kshitij Singhi martinian.ksin...@gmail.com Date: Mon, Oct 10, 2011 at 7:36 PM Subject: Re: [OSL | CCIE_Voice] Fractional MGCP To: Ken Wyan kew...@gmail.com Cc: ccie_voice@onlinestudylist.com Attached are the screenshots of the GW configuration (main page and endpoint). Site A and Site B are more or less identical (except for the domain names) and both have the defaults configured (SF set to 4 and Display IE delivery, Redirecting IE delivery outbound/inbound being checked). I performed the following tests: 1. Maxing out the channels and then checking if it fails over to the next option in the RG. 2. Maxing out the channels and then checking if it fails over to the next option in the RL. 3. Maxing out the channels by making incoming calls and then calling out to check if the call goes out via the next option in the RG. 4. Calls during SRST. (both incoming and outgoing to see if there is any change in behavior) 5. Bringing the Site out of SRST to see if there is any change in behavior. FOR POINT 1 GIVEN ABOVE Call 1 going out via channel 3 == Oct 10 12:57:30.789: MGCP Packet received from 192.168.10.47:2427--- CRCX 463 S2/DS1-0/3...@site-b.yourdomain.com MGCP 0.1 C: D202e65000F50003 X: 3 L: p:20, a:PCMU, s:off, t:b8, fxr/fx:t38 M: recvonly R: D/[0-9ABCD*#] Q: process,loop --- Oct 10 12:57:30.813: ISDN Se2/0:23 Q931: TX - SETUP pd = 8 callref = 0x0003 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98383 Exclusive, Channel 3 Calling Party Number i = 0x0081, '3002' Plan:Unknown, Type:Unknown Called Party Number i = 0x80, '911' Plan:Unknown, Type:Unknown Oct 10 12:57:30.829: ISDN Se2/0:23 Q931: RX - CALL_PROC pd = 8 callref = 0x8003 Channel ID i = 0xA98383 Exclusive, Channel 3 Call 2 going out via channel 2 === Oct 10 12:57:39.806: MGCP Packet received from 192.168.10.47:2427--- CRCX 467 S2/DS1-0/2...@site-b.yourdomain.com MGCP 0.1 C: D202e65300F50004 X: 2 L: p:20, a:PCMU, s:off, t:b8, fxr/fx:t38 M: recvonly R: D/[0-9ABCD*#] Q: process,loop --- Oct 10 12:57:39.826: ISDN Se2/0:23 Q931: TX - SETUP pd = 8 callref = 0x0004 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98382 Exclusive, Channel 2 Calling Party Number i = 0x0081, '3002' Plan:Unknown, Type:Unknown Called Party Number i = 0x80, '911' Plan:Unknown, Type:Unknown Oct 10 12:57:39.842: ISDN Se2/0:23 Q931: RX - CALL_PROC pd = 8 callref = 0x8004 Channel ID i = 0xA98382 Exclusive, Channel 2 Call 3 going out via channel 1 == Oct 10 12:57:49.279: MGCP Packet received from 192.168.10.47:2427--- CRCX 471 S2/DS1-0/1...@site-b.yourdomain.com MGCP 0.1 C: D202e65600F50005 X: 1 L: p:20, a:PCMU, s:off, t:b8, fxr/fx:t38 M: recvonly R: D/[0-9ABCD*#] Q: process,loop --- Oct 10 12:57:49.299: ISDN Se2/0:23 Q931: TX - SETUP pd = 8 callref = 0x0005 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i
Re: [OSL | CCIE_Voice] [UCCX7-UCCX_IPIVR_7_0_1.iso on Windows 2003 R2 Enterprise]
hey Michael , in fact the CCX installation is a bit more advanced than the ccm , the windows should be Cisco Published windows and you have to install SQL as well and then the ccx , yes there is some registery to change to make it working on VM , i will try to get them for you soon (in the next 2 days ) as long as with full explanation Ash On Sat, Oct 8, 2011 at 5:30 PM, michael.se...@compucom.com wrote: Greetings, I'm trying to install UCCX7-UCCX_IPIVR_7_0_1.iso on Windows 2003R2 Enterprise. I've been told that it will work with certain modifications, registry hacks and the like. I'm trying to find out the details of how to go about doing this and wanted to run it by the group. Any information would be appreciated since this is my third install and hopefully the last. Another question I have is do you do the complete SQL installation. Is there a link out there that explains all this that I'm missing. Installing on ESXI VMWARE VM. Thank you --ms ___ 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] Accessing CUCM from remote site
Hey Nish , the info you provided is not enough , can you try to ping from the ccm to the SVI of Site B ? can you make sure that you have the SVI interface state as up up ? can you please make more advanced troubleshooting and share it result with us , note we dont have access to your system to find out this by ourself so your collaboration will help us more Ash On Sun, Oct 9, 2011 at 1:35 AM, Nish Tarpara nishi...@hotmail.com wrote: Hi All, I am having problem accessing CUCM(on VMware workstation6.5) from Site B and able to connect fine from HQ site(Vlan10,20,30) and also able to ping CUCM server via switch. I am using same switch for Site B to connect Site B routers and phone with different Vlans 40 50. from site B router i am able to ping VM Workstation PC but not the CUCM server which reside on that VM. Please advise soon as i am unable to work further. Thank you, Nish ___ 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] CUCM7 with SIP Providers
yeah sure , can you please let me know if your provider need username and PWD , realm ? if not you just need to go ahead and create sip trunk pointing to the ip of your provider , if not please let me know so i can share the steps to make the ccm authenticate with your sip provider Ash On Sun, Oct 9, 2011 at 12:00 PM, Azher Mughal az...@hep.caltech.edu wrote: Hi, I am trying to configure CUCM7 with a SIP Service provider Voip.ms. It works fine with Asterisk. Anyone can give steps to follow ? Thanks -Azher ___ 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] CUCM7 with SIP Providers
okay i don't fully understand what we need to do now , with or without username and realem ? do you want to configure the ccm with sip trunk direct to the SIP ITSP or you want to configure CUBE in between ? if you any Authentication with your provider , you have to configure the Enterprise parameter of Cluster ID This parameter supports Cisco Unified Communications Manager challenges to the identity of the SIP user agent that is sending a SIP request on the SIP trunk. and you have to configure Realm from the User management , and you have to enabled the Authentication on the SIP trunk security profile associated with the SIP Trunk and you have to configure application user of what you got from the provider , in fact making the ccm authenticate with the ITSP is not a good Idea , i think having CUBE in between will be much better in term of scalability Ash On Sun, Oct 9, 2011 at 7:09 PM, Azher Mughal az...@hep.caltech.edu wrote: Voip.ms require username, password and any realm works. For asterisk I am using insecure=very along with peer ip of CUCM so CUCM don't need to do authentication on the trunk. Thanks -Azher On 10/9/2011 12:33 PM, Ashraf Ayyash wrote: yeah sure , can you please let me know if your provider need username and PWD , realm ? if not you just need to go ahead and create sip trunk pointing to the ip of your provider , if not please let me know so i can share the steps to make the ccm authenticate with your sip provider Ash On Sun, Oct 9, 2011 at 12:00 PM, Azher Mughal az...@hep.caltech.edu wrote: Hi, I am trying to configure CUCM7 with a SIP Service provider Voip.ms. It works fine with Asterisk. Anyone can give steps to follow ? Thanks -Azher ___ 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] CUCM7 with SIP Providers
forget to add , the ccm will not register with any ITSP , the Above config is to make the ccm accept the Authentication Challange , if your provider require registration then you have to get CUBE between the ccm and the provider to satify his requirement if any registration is required Note also some provider work based on IP add and port , so you will not need to have any Authentication config . Ash On Sun, Oct 9, 2011 at 10:30 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote: okay i don't fully understand what we need to do now , with or without username and realem ? do you want to configure the ccm with sip trunk direct to the SIP ITSP or you want to configure CUBE in between ? if you any Authentication with your provider , you have to configure the Enterprise parameter of Cluster ID This parameter supports Cisco Unified Communications Manager challenges to the identity of the SIP user agent that is sending a SIP request on the SIP trunk. and you have to configure Realm from the User management , and you have to enabled the Authentication on the SIP trunk security profile associated with the SIP Trunk and you have to configure application user of what you got from the provider , in fact making the ccm authenticate with the ITSP is not a good Idea , i think having CUBE in between will be much better in term of scalability Ash On Sun, Oct 9, 2011 at 7:09 PM, Azher Mughal az...@hep.caltech.edu wrote: Voip.ms require username, password and any realm works. For asterisk I am using insecure=very along with peer ip of CUCM so CUCM don't need to do authentication on the trunk. Thanks -Azher On 10/9/2011 12:33 PM, Ashraf Ayyash wrote: yeah sure , can you please let me know if your provider need username and PWD , realm ? if not you just need to go ahead and create sip trunk pointing to the ip of your provider , if not please let me know so i can share the steps to make the ccm authenticate with your sip provider Ash On Sun, Oct 9, 2011 at 12:00 PM, Azher Mughal az...@hep.caltech.edu wrote: Hi, I am trying to configure CUCM7 with a SIP Service provider Voip.ms. It works fine with Asterisk. Anyone can give steps to follow ? Thanks -Azher ___ 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] Why not set CSS in the Device Pool?
yes you right , the exam is not to test scalability but in real live this option on the dp is very usefull and in case of aar , the aar group will work for everything on the dp level but the line , and this is a huge time saver in case of aar , Ash On Sat, Oct 8, 2011 at 11:55 AM, Mark Reed marklr...@gmail.com wrote: I never see this done for some reason. With the number of phones were talking about it isn't a huge time saver, but I'll take every second I can get at this point. I did my entire mock lab this way yesterday and everything worked great except the aar-group which I still needed on the line itself. But everything worked exactly like I had set it per device. Am I missing something or am I just out of the loop and people are doing it this way? -- Thanks, Mark L Reed ___ 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] Fractional MGCP
hello All , well, i apologize if i got off track in disccussing this issue in the alias ( this apology including your Kshitji ) i think ccie is getting me aggressive anyway , i reason you cannot find this workaround in any of cisco doc is the fact that we dont support this feature , mgcp is not desinged to work in fraction connection however cisco have interduce this feature because mgcp is more prefered for the customer as its very easy to setup , i worked on a very heavy mgcp case in the past cauing me to read the whole rfc of the mgcp and i i was in touch with the TAC expert and the DE in charge of this feature and the discussion ended to say that TAC doesnt support fraction mgcp and this is a temp workaround you can use in the time being tpo avoid cal failure when the ccm will setup call on a non-used bchannel and this feature is under study for feature full suppor on the ccm nativly but we dont have any estimated release or time yet , Thanks Ash On Sat, Oct 8, 2011 at 2:00 PM, Ken Wyan kew...@gmail.com wrote: Hi Kshitij, Logically it should use next GW in RG ( to next RG , etc..) when all 3 channels are full in first GW. (As per your obsevations it should be) But better to test as at times CUCM server behaves very strange. In fact a TAC Engineer ( from India ) told me to use this service parameter to support fractional MGCP (when I opened a TAC case for fractional E1 in MGCP long time back). Cisco docs never say to use this service parameter for fractional E1/T1 MGCP it is for temporary busy-out of channels (maintenance purposes). I guess a TAC expert has guided this way to overcome a bug in a particular code or to give a quick solution for fractional MGCP ( rather than time-consuming manual MGCP configuration) also not to affect cisco's PVDM sales volume. Thank you for your findings if Ash can check again this with TAC experts it would be very nice. Ken On Sat, Oct 8, 2011 at 4:03 PM, Kshitij Singhi martinian.ksin...@gmail.com wrote: Hmm... will max out my MGCP channels on Monday and check if calls move out of the backup endpoint configured in the RG/RL. Not sure if I tested this when I was practicing but as far as I remember, I have. Will update soon!!! ___ 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] Fractional MGCP
ha! i have tested/studied this away long time ago and i have also consulted TAC expert to confirm my tests ( as i accept other people thought and knowledge) and i fully understand the mgcp signalling longtime ago and got alot of case for the very same subject we are discussing now , thats why i am sure and thats why i am telling you that you are publishing a complete wrong info !! take this chat off from the alias now , people start to be confused ...and you keep misleading the same info ! Ash On Fri, Oct 7, 2011 at 1:38 AM, Kshitij Singhi martinian.ksin...@gmail.com wrote: LOL. The tests/debugs/show outputs clarify everything - please dig deeper and try this out in this way the next time you sit to practice. You will be surprised. On Fri, Oct 7, 2011 at 1:04 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote: Kshitij , oh again ! the Isdn layer is lower than the mgcp , if you will busy out the channel the ccm will not consider them OOS , the 500 unknown endpoint is not enough to make the ccm busy out the non-used b-channel , try to busy them out from the service parameters and you will see the same exact debugs output you used them , in order to busy them out you have to use the service parameters to do this for you , otherwise the ccm will send setup call to those channels and you will see circuit unavailable coming back as a replay !! what ccm version you are using ? 7.0 ? Ash On Fri, Oct 7, 2011 at 1:22 AM, Kshitij Singhi martinian.ksin...@gmail.com wrote: Noticed 2 modifications that can be made - it should be spelled niece And status 2 = idle (which effectively could mean that it is not in use i.e. there isn't an active call on it). I was thinking from the perspective of Not in use as in it's not participating in call routing. On Fri, Oct 7, 2011 at 12:36 PM, Kshitij Singhi martinian.ksin...@gmail.com wrote: OK. Let's dance. Given below is my configuration (the pertinent section): show run | sec controller controller T1 0/0/0 framing esf linecode b8zs pri-group timeslots 1-3,24 service mgcp show run | sec interface Serial0/0/0 interface Serial0/0/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn bind-l3 ccm-manager isdn outgoing display-ie isdn outgoing ie redirecting-number no cdp enable show run | in mgcp pri-group timeslots 1-3,24 service mgcp ccm-manager mgcp mgcp mgcp call-agent 192.168.10.47 service-type mgcp version 0.1 no mgcp package-capability res-package no mgcp timer receive-rtcp mgcp bind control source-interface GigabitEthernet0/0.102 mgcp bind media source-interface GigabitEthernet0/0.102 mgcp profile default show run | in ccm isdn bind-l3 ccm-manager ccm-manager switchback immediate ccm-manager redundant-host 192.168.10.46 ccm-manager mgcp no ccm-manager fax protocol cisco ccm-manager music-on-hold do show ccm-manager MGCP Domain Name: SiteA Priority Status Host Primary Registered 192.168.10.47 First Backup Backup Ready 192.168.10.46 Second Backup None Current active Call Manager: 192.168.10.47 Backhaul/Redundant link port: 2428 Failover Interval: 30 seconds Keepalive Interval: 15 seconds Last keepalive sent: 21:50:15 PDT Oct 6 2011 (elapsed time: 00:00:10) Last MGCP traffic time: 21:50:15 PDT Oct 6 2011 (elapsed time: 00:00:10) Last failover time: 01:07:35 PDT Oct 1 2011 from (192.168.10.47) Last switchback time: 01:08:05 PDT Oct 1 2011 from (192.168.10.46) Switchback mode: Immediate MGCP Fallback mode: Not Selected Last MGCP Fallback start time: None Last MGCP Fallback end time: None MGCP Download Tones: Disabled TFTP retry count to shut Ports: 2 Backhaul Link info: Link Protocol: TCP Remote Port Number: 2428 Remote IP Address: 192.168.10.47 Current Link State: OPEN Statistics: Packets recvd: 1 Recv failures: 0 Packets xmitted: 1 Xmit failures: 0 PRI Ports being backhauled: Slot 0, VIC 0, port 0 FAX mode: disable Configuration Error History: Let's take a look at this section in point 1: we here talking about the B Channel not the D-Channal so getting 500 on the AUEP doesnt mean the mgcp gw will busy out this channel and thats exactly why we have this service paramert in the ccm to busy out the b-chann Since I have only 3 channels configured on the T1 controller, I took a debug mgcp packet and saw: Oct 7 04:48:00.453: MGCP Packet sent to 192.168.10.47:2427--- RSIP 696986311 *@SiteA MGCP 0.1 RM: restart --- Oct 7 04:48:00.457: MGCP Packet received from 192.168.10.47:2427--- 200 696986311
Re: [OSL | CCIE_Voice] VoiceView Under SRST
hey man the voice view cannot work during the SRST , i mean you cannot preserve it , you will see the inbonx shoing when you logged in but you cannot do any further tasks , i hope this is clear Ash On Fri, Oct 7, 2011 at 7:27 AM, amit batra batraji...@yahoo.com wrote: 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 ___ 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] Voiceview Express: Phone authentication is not working - How to debug this? -
the command i mentioned is CUE commands and you have to add them with CME -CUE integration , can you please giv them a go and let us know the results Ash On Fri, Oct 7, 2011 at 3:47 AM, Robert Schuknecht rschukne...@gmx.de wrote: I am using CME as my callagent, so CUE is integrated with CME via SIP. /Robert Von: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] Im Auftrag von Shrini Gesendet: Freitag, 7. Oktober 2011 00:46 An: ccie_voice@onlinestudylist.com Betreff: Re: [OSL | CCIE_Voice] Voiceview Express: Phone authentication is not working - How to debug this? - To resolve this issue, Assign phones to JTAPI user configured for CUE. On 10/6/2011 2:07 PM, Robert Schuknecht wrote: Hi, Today I tried to get Voiceview Express working, without luck. As far as I can see in sniffer traces the Phone authentication is not working. And now I need some help to find my error. I already read the archive and the CUE Admin Guide, but I am not able to find the right solution. Any help is really really appreciated! What I did so far: - Used CUE for phone authentication (url authentication http://10.1.137.10/voiceview/authentication/authenticate.do) with fallback authentication url (http://10.1.137.1/CCMCIP/authenticate.asp) - Used the command authentication credential admin cisco under telephony-service - Searched the cisco supportforums and the Bug-Toolkit but I did not find any helpful My configurations: CME: R3#sh run | sec telephony-service telephony-service no auto-reg-ephone em logout 19:0 23:0 7:0 max-ephones 10 max-dn 10 no-reg both ip source-address 10.1.137.1 port 2000 service phone webAccess 0 system message Your current options url services http://10.1.137.10/voiceview/common/login.do VoiceView Express url authentication http://10.1.137.1/CCMCIP/authenticate.asp cnf-file perphone load 7961 SCCP41.8-3-3S time-zone 23 time-format 24 date-format dd-mm-yy voicemail 3600 max-conferences 8 gain -6 call-forward pattern .T moh music-on-hold.au web admin system name admin password cisco dn-webedit time-webedit transfer-system full-consult create cnf-files version-stamp 7960 Oct 06 2011 22:19:49 ephone-dn 1 octo-line number 3001 no-reg both description 3214-3001 name SITEC_PHONE_1 call-forward all 3600 call-forward busy 3600 call-forward noan 3600 timeout 10 ephone 1 device-security-mode none mac-address 0017.59E9.6A80 ephone-template 1 max-calls-per-button 5 busy-trigger-per-button 1 username scphn1 password cisco type 7961 button 1:1 CUE: site name local phone-authentication username admin password cisco site-hostname 10.1.137.1 web web username admin password cisco end site service phone-authentication end phone-authentication service voiceview enable end voiceview Used Software Versions: CUE: se-10-1-137-10# show software versions Cisco Unity Express version (7.0.1) Technical Support: http://www.cisco.com/techsupport Copyright (c) 1986-2008 by Cisco Systems, Inc. Components: - CUE Voicemail Language Support version 7.0.1.0 se-10-1-137-10# show software licenses Installed license files: - voicemail_lic.sig : 12 MAILBOX LICENSE - ivr_lic.sig : 8 PORT IVR BASE LICENSE Core: - Application mode: CCME - Total usable system ports: 24 Voicemail/Auto Attendant: - Max system mailbox capacity time: 18000 - Default # of general delivery mailboxes: 5 - Default # of personal mailboxes: 12 - Max # of configurable mailboxes: 17 Interactive Voice Response: - Max # of IVR sessions: 8 Languages: - Max installed languages: 5 - Max enabled languages: 5 se-10-1-137-10# CME: c2800nm-adventerprisek9_ivs-mz.124-22.T2.bin Phone (7961): SCCP-41.8-3-3S /Robert ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Fractional MGCP
i have gave explanation why your info's is wrong and there is no etiquette in the network , its either working or not working , true or false , i am always giving a prove from real labs and i never used the company that i work for to give people reason to take the info i shared as trusted and this is not poolshitting ,, the only poolshitting is to come and say because i am working for Cisco TAC my info must be trusted and people have no right to say/ prove the opposite go ahead and speak with anyone from the real expert / escalation team and they will tell you if your info is right or wrong even though i don't care , i only email the alias because this can be very likely a question in the exam and then people will follow MR Cisco TAC engineer who share a wrong info and then they will get a bad score on the GW section , even though you always INSULT back when you answering , i really don't pay Shxit to your replays , stop share non-tested info and verify your answer before answer it and you will never see me replaying for correct info saying its wrong info , Be professional please and keep in mind that Next time i will not accept any stupid word back from your side i will carry it to your managment straight away and we will see if your contract have Cisco Employee NDA commitment or not Ash On Wed, Oct 5, 2011 at 10:40 PM, Kshitij Singhi martinian.ksin...@gmail.com wrote: It's not wrong and you desperately need to stop bullshi**ing. I know precisely what I am allowed/not allowed to do and you are no one and will always be no one to tell me or anyone about it. Follow it if you want to, ignore it if you believe you know better - either way all the best for your exam. Things are only as complicated as you make them - just a tip for life. Instead of arguing on a public forum and making such resentful and rude statements, please ping me directly if you have any issues and prove me wrong - I will definitely rescind any statements made by me that have been proven wrong conclusively. PLEASE don't stoop lower than this. PLEASE take this off the OSL. I'm literally begging you - PLEASE. Once again, read the etiquette section thoroughly. On Thu, Oct 6, 2011 at 9:16 AM, Ashraf Ayyash ash.ayy...@gmail.com wrote: this is completly wrong Kshitij , 1- the mgcp layer have nothing to do with the isdn layer even though the l3 is binded to the ccm , we here talking about the B Channel not the D-Channal so getting 500 on the AUEP doesnt mean the mgcp gw will busy out this channel and thats exactly why we have this service paramert in the ccm to busy out the b-chann and after that you can verify this from the show perf query class of the mgcp pri and you will see the bchannl not in use on status 2 . 2- in term of the ccie scope , this is also completey wrong , if you have mgcp gw question and you have been asked to use on certain number of b-chann , what do you think they are asking you to do pri group command and move on with 4 points ? or cisco doest have enough dsp to put on the router exam ? try do show invent and you will see what is loaded on the exam router . finally please note that you are using / talking by the name of Cisco TAC , even though you are not allowed do so , however at least be more accurate on the answer you are publising here as now you bring TAC to the game and this is not good for the TAC picture on the public aliases . the answer of this question is , certainly they will need you to use the service paramater to busy out the unused channel , note also that this is a ccie level exam so i would suggest that you always try to find out why they asking any easy question because you will find that this easy question is just a pointer to do something deeper which will give the ccie program a chance to test your knowledge on a CCIE Level . Ash On Sun, Oct 2, 2011 at 10:53 PM, Kshitij Singhi martinian.ksin...@gmail.com wrote: Just to add, from a CCIE scope, the IPX way is enough - we simply need to manually add the pri-group timeslots command with the correct number of channels that need to be used. Not required to modify any service parameter on CUCM. From the exams perspective, I would suggest: 1. Downloading the configuration from CUCM by adding the ccm-manager config and ccm-manager config server commands after configuring everything on CUCM. 2. This should add the pri-group timeslots command with 24 channels. 3. Shut down the voice-port/serial interface/controller and remove L3 binding from the Serial interface 4. Remove the ccm-manager config/ccm-manager config server commands. 5. Remove the pri-g timeslots command and re-add it with the correct number of channels. No shut the controller and the serial interface (if applicable). 6. Manually add L3 binding on the Serial interface. Issue a no mgcp/mgcp Should be good to go. On Mon, Oct 3, 2011 at 11:17 AM, Kshitij Singhi martinian.ksin
Re: [OSL | CCIE_Voice] Voiceview Express: Phone authentication is not working - How to debug this? -
try those commands on the CUE site name local phone-authentication username username password password and check this DOC : www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/administrator/AA_and_VM/guide/vview.html Ash On Thu, Oct 6, 2011 at 7:32 PM, Ray jonha...@yahoo.com wrote: make sure the address u put in the http is pointing to cue as below. cue ip is 10.50.10.124. also add a phone service with this url : http://10.50.10.125/voiceview/common/login.do and subcribe it to the phones also add the phone to the cue application user that u create in cucm. i hope this helps. let me know how you make it. #telephony-service cme(config-telephony)#url services http://10.50.10.125/voiceview/common/login.do cme(config-telephony)#url authentication http://10.50.10.125/voiceview/authentication/authenticate.do From: Shrini linuxbos...@gmail.com To: ccie_voice@onlinestudylist.com Sent: Thursday, October 6, 2011 5:46 PM Subject: Re: [OSL | CCIE_Voice] Voiceview Express: Phone authentication is not working - How to debug this? - To resolve this issue, Assign phones to JTAPI user configured for CUE. On 10/6/2011 2:07 PM, Robert Schuknecht wrote: Hi, Today I tried to get Voiceview Express working, without luck. As far as I can see in sniffer traces the Phone authentication is not working. And now I need some help to find my error. I already read the archive and the CUE Admin Guide, but I am not able to find the right solution. Any help is really really appreciated! What I did so far: - Used CUE for phone authentication (url authentication http://10.1.137.10/voiceview/authentication/authenticate.do) with fallback authentication url (http://10.1.137.1/CCMCIP/authenticate.asp) - Used the command authentication credential admin cisco under telephony-service - Searched the cisco supportforums and the Bug-Toolkit but I did not find any helpful My configurations: CME: R3#sh run | sec telephony-service telephony-service no auto-reg-ephone em logout 19:0 23:0 7:0 max-ephones 10 max-dn 10 no-reg both ip source-address 10.1.137.1 port 2000 service phone webAccess 0 system message Your current options url services http://10.1.137.10/voiceview/common/login.do VoiceView Express url authentication http://10.1.137.1/CCMCIP/authenticate.asp cnf-file perphone load 7961 SCCP41.8-3-3S time-zone 23 time-format 24 date-format dd-mm-yy voicemail 3600 max-conferences 8 gain -6 call-forward pattern .T moh music-on-hold.au web admin system name admin password cisco dn-webedit time-webedit transfer-system full-consult create cnf-files version-stamp 7960 Oct 06 2011 22:19:49 ephone-dn 1 octo-line number 3001 no-reg both description 3214-3001 name SITEC_PHONE_1 call-forward all 3600 call-forward busy 3600 call-forward noan 3600 timeout 10 ephone 1 device-security-mode none mac-address 0017.59E9.6A80 ephone-template 1 max-calls-per-button 5 busy-trigger-per-button 1 username scphn1 password cisco type 7961 button 1:1 CUE: site name local phone-authentication username admin password cisco site-hostname 10.1.137.1 web web username admin password cisco end site service phone-authentication end phone-authentication service voiceview enable end voiceview Used Software Versions: CUE: se-10-1-137-10# show software versions Cisco Unity Express version (7.0.1) Technical Support: http://www.cisco.com/techsupport Copyright (c) 1986-2008 by Cisco Systems, Inc. Components: - CUE Voicemail Language Support version 7.0.1.0 se-10-1-137-10# show software licenses Installed license files: - voicemail_lic.sig : 12 MAILBOX LICENSE - ivr_lic.sig : 8 PORT IVR BASE LICENSE Core: - Application mode: CCME - Total usable system ports: 24 Voicemail/Auto Attendant: - Max system mailbox capacity time: 18000 - Default # of general delivery mailboxes: 5 - Default # of personal mailboxes: 12 - Max # of configurable mailboxes: 17 Interactive Voice Response: - Max # of IVR sessions: 8 Languages: - Max installed languages: 5 - Max enabled languages: 5 se-10-1-137-10# CME: c2800nm-adventerprisek9_ivs-mz.124-22.T2.bin Phone (7961): SCCP-41.8-3-3S /Robert ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Call-Forward to VM
check what vm profile is assigned to this number , if its the default go ahead and specify the vm profile manually ash On Wed, Oct 5, 2011 at 3:24 PM, Jason Lee jas7...@gmail.com wrote: All, Having a weird problem. I have CUC integrated with CUCM via SCCP. I'm able to access the CUC server by dialing the VM pilot or pressing the messages button on the phone. When I forward calls to VM under line configuration using the VM checkbox I get a fast-busy. If I uncheck the box and manually enter the VM pilot number it works fine. Has anyone ever run into this problem? thanks, Jason ___ 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] Fractional MGCP
this is completly wrong Kshitij , 1- the mgcp layer have nothing to do with the isdn layer even though the l3 is binded to the ccm , we here talking about the B Channel not the D-Channal so getting 500 on the AUEP doesnt mean the mgcp gw will busy out this channel and thats exactly why we have this service paramert in the ccm to busy out the b-chann and after that you can verify this from the show perf query class of the mgcp pri and you will see the bchannl not in use on status 2 . 2- in term of the ccie scope , this is also completey wrong , if you have mgcp gw question and you have been asked to use on certain number of b-chann , what do you think they are asking you to do pri group command and move on with 4 points ? or cisco doest have enough dsp to put on the router exam ? try do show invent and you will see what is loaded on the exam router . finally please note that you are using / talking by the name of Cisco TAC , even though you are not allowed do so , however at least be more accurate on the answer you are publising here as now you bring TAC to the game and this is not good for the TAC picture on the public aliases . the answer of this question is , certainly they will need you to use the service paramater to busy out the unused channel , note also that this is a ccie level exam so i would suggest that you always try to find out why they asking any easy question because you will find that this easy question is just a pointer to do something deeper which will give the ccie program a chance to test your knowledge on a CCIE Level . Ash On Sun, Oct 2, 2011 at 10:53 PM, Kshitij Singhi martinian.ksin...@gmail.com wrote: Just to add, from a CCIE scope, the IPX way is enough - we simply need to manually add the pri-group timeslots command with the correct number of channels that need to be used. Not required to modify any service parameter on CUCM. From the exams perspective, I would suggest: 1. Downloading the configuration from CUCM by adding the ccm-manager config and ccm-manager config server commands after configuring everything on CUCM. 2. This should add the pri-group timeslots command with 24 channels. 3. Shut down the voice-port/serial interface/controller and remove L3 binding from the Serial interface 4. Remove the ccm-manager config/ccm-manager config server commands. 5. Remove the pri-g timeslots command and re-add it with the correct number of channels. No shut the controller and the serial interface (if applicable). 6. Manually add L3 binding on the Serial interface. Issue a no mgcp/mgcp Should be good to go. On Mon, Oct 3, 2011 at 11:17 AM, Kshitij Singhi martinian.ksin...@gmail.com wrote: Fractional MGCP controlled PRIs are not supported by TAC. It's not possible to configure a fractional PRI by downloading the config from CUCM via the following commands: ccm-manager config ccm-manager config server IP However, a fractional MGCP controlled PRI works fine when the GW is manually configured. To do this, we need to add the following commands on the GW: ccm-m mgc ccm-m call-agent IP ccm-m redun IP (If applicable) controller t1 x/y/z pri-g time 1-5 ser mgc (assuming 5 channels are being used - the Telco will need to be configured accordingly as well) int ser x/y/z:23 isdn bind-l3 ccm-manager mgcp The statement CUCM does not support a fractional MGCP controlled PRI might not be entirely accurate since CUCM definitely works great with a fractional MGCP controlled PRI. I guess saying that CUCM cannot auto configure a fractional MGCP controlled PRI would be more accurate. In the case of a fractional PRI, CUCM sends AUEP messages to the GW for the unconfigured channels on the PRI, but the GW responds with an endpoint unknown message - hence, CUCM does not consider those channels during call routing. On Sun, Oct 2, 2011 at 11:42 AM, ccie_voice-requ...@onlinestudylist.com wrote: Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: I need hardware Vpn assistance on a session now (Marko Milivojevic) 2. Re: I need hardware Vpn assistance on a session now (pithog...@yahoo.com) 3. Re: Fractional MGCP (Robert Thomas) 4. Re: I need hardware Vpn assistance on a session now (Rrcrumm) 5. PREDOT DDI vs NANP:PREDOT DDI (Ken Wyan) -- Message: 1 Date: Sat, 1 Oct 2011 13:51:57 -0700 From: Marko Milivojevic mar...@ipexpert.com To: edgar feliz
Re: [OSL | CCIE_Voice] Gatekeeper configuration
Guys , the h323 bind command is not a GK command , its for the H323 GW so you can have GK on interface and have the normal config of the GK and the H323 bind under the interface which will work as H323 GW with the ccm H323 bind can be usefull if you have Q said bind all media and signalling to specific interface IP , h323 bind in this case will do RTP binding for you , i hope this is clear Ash On Tue, Oct 4, 2011 at 8:45 AM, Rrcrumm rrcr...@yahoo.com wrote: Did you do a gateway in config mode on the branch rtr? Is it registered? Randall Sent from my iPhone On Oct 4, 2011, at 1:12 AM, darshan ccievoice0...@hotmail.com wrote: Dear All; I have 2 requirements and I don’t know which one is the correct configuration.. 1. For SIte C CME gateway Configure SIteC Router as H323 gateway. Make sure that all inbound outbound H323 traffic is sourced from the local interface 144.102.66.254/24 2.In Gatekeeper another Question is there SIte C should use its Loopback address for all communication with the gatekeeper. My configuration is Sc#interface loopback 0 desc SC Loopback h323-gateway voip interface ip address 144.1.66.254 255.255.255.0 h323-gateway voip interface h323-gateway voip id GK ipaddr 144.1.64.254 1719 h323-gateway voip bind srcaddr 144.1.66.254 OR Sc#interface vlan502 desc SC Voice h323-voip interface ip address 144.102.66.254 255.255.255.0 h323-gateway voip interface h323-gateway voip id GK ipaddr 144.1.64.254 1719 h323-gateway voip bind srcaddr 144.102.66.254 144.1.66.254--- is the SIte C Loopback address 144.102.66.254-- is the Site C Voice Vlan 502 Appreciate to help me in this regard.. reagrds Tashu ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Trust command
because you added the command mls qos trust device cisco-phone the switch will know via CDP that you are connecting Cisco Phone other wise the default behavior (*after enable the mls qos on the switch) will be applied of not trust , i would say that 99% you will be trusting the COS on the switch port connected to the phones and the DSCP of the port connected to the servers . Ash On Sat, Oct 1, 2011 at 3:29 AM, Nowork_onlyfun noworkonlyf...@gmail.com wrote: Hi Guys Regarding the mls qos trust command. If the question demands to trust dscp values. On the interface connected to the phone. If I configure mls qos trust dscp and mls qos trust device cisco-phone. Does this means that the interface will trust the incoming dscp only if it's coming from a cisco phone ? Or it has detected cisco phone on the interface ? Or will it be trusting the dscp regardless of connected device ? Because of mls qos trust dscp command ? Thanks. Sent from my iPad ___ 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] Fractional MGCP
the correct answer for your Question is to busy out the non-defind B channels from the Service parameters , Note that CCM doesn't support Fractional T1/E1 over mgcp , this service parameter is kind of workaround and i will use it myself if will setup Fractional T1/E1 over mgcp , Ash In IPX workbooks ; they limit number of channels just by pri-group timeslots 1-5,24 service mgcp command only. ( I didn't go through all the solutions yet , but I have seen this few times so far) But I think fractional pri for MGCP is not supported by CUCM as per Cisco documentation. Somehow there's a way as below ( B chan maintenance status poll ). https://supportforums.cisco.com/thread/97578 ( not an official cisco document) For CCIE scope , does IPX way is enough? But it will not do the required job. Wyan ___ 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] lab3 gatekeeper
81 mean unallocated unassigned number , you dial the wrong number check this DOC : http://www.cisco.com/en/US/tech/tk801/tk379/technologies_tech_note09186a008012e95f.shtml Thanks Ash On Thu, Sep 29, 2011 at 8:37 AM, Ray jonha...@yahoo.com wrote: what option or command do i have to take out inorder to the the q931 cause code id 8081.. because i set it up and confired all the commands in the workbook and everything is working fine..i can't generate the cause code 8081.. HQ# gatkeeper zone local GK ccievoice.com 142.1.64.254 zone remote BBGK cisco.com 155.26.1.100 1719 zone prefix BBGK 01144* zone prefix BBGK 44* no shut ___ 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] lab3 gatekeeper
i dont fully understand what you are looking for , on the ccm traces you will find Cause = the disconnect cause and its Q931 in there few line above the release complete message , and to know what is the cause you need to use the DOC i have sent to you , which will explain and show you what does it mean , if you have more specific Q please let me /us know Ash On Thu, Sep 29, 2011 at 12:39 PM, Ray jonha...@yahoo.com wrote: what option or command do i have to take out/uncheck inorder to see the q931 cause code id 8081 in the CUCM traces.. because i set it up and confirmed all the commands in the workbook and everything is working fine..i can't generate the cause code 8081..Anyone knows how to create the q931 cause code id 8081? From: Ray jonha...@yahoo.com To: ccie voice ccie_voice@onlinestudylist.com Sent: Thursday, September 29, 2011 10:37 AM Subject: [OSL | CCIE_Voice] lab3 gatekeeper what option or command do i have to take out inorder to the the q931 cause code id 8081.. because i set it up and confired all the commands in the workbook and everything is working fine..i can't generate the cause code 8081.. HQ# gatkeeper zone local GK ccievoice.com 142.1.64.254 zone remote BBGK cisco.com 155.26.1.100 1719 zone prefix BBGK 01144* zone prefix BBGK 44* no shut ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] lab3 gatekeeper
i have recreate it in my lab to be accurate , in my config i made the
incoming dialpeer with G711 and i am sending G729 call :
loneClusterNID::10.10.210.11CT::2,100,16,10.2IP::10.10.100.2DEV::Port
15861LVL::DetailedMASK::0100
09/29/2011 22:32:58.497 CCM|value H323-UserInformation ::=
|CLID::StandAloneClusterNID::10.10.210.11LVL::State
TransitionMASK::0100
09/29/2011 22:32:58.498 CCM|{
h323-uu-pdu
{
h323-message-body releaseComplete :
{
protocolIdentifier { 0 0 8 2250 0 5 },
callIdentifier
{
guid '80D4BC7E562A51E8060013020A0AC91D'H
}
},
h245Tunneling FALSE
}|CLID::StandAloneClusterNID::10.10.210.11LVL::State
TransitionMASK::0100
09/29/2011 22:32:58.498
CCM|}|CLID::StandAloneClusterNID::10.10.210.11LVL::State
TransitionMASK::0100
09/29/2011 22:32:58.498 CCM|
|CLID::StandAloneClusterNID::10.10.210.11LVL::State
TransitionMASK::0040
09/29/2011 22:32:58.498 CCM|Out Message -- H225ReleaseCompleteMsg --
Protocol=
H225Protocol|CLID::StandAloneClusterNID::10.10.210.11LVL::SignificantMASK::0040
09/29/2011 22:32:58.498 CCM|Ie - Q931CauseIe IEData= 08 02 80 AF
|CLID::StandAloneClusterNID::10.10.210.11LVL::State
TransitionMASK::0040
Q931CauseIe IEData= 08 02 80 AF check the doc i have pointed and you
will find that AF is resources unavailable ...
i hope this is clear
Ash
On Thu, Sep 29, 2011 at 2:58 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote:
i dont fully understand what you are looking for ,
on the ccm traces you will find Cause = the disconnect cause and
its Q931 in there few line above the release complete message ,
and to know what is the cause you need to use the DOC i have sent to
you , which will explain and show you what does it mean ,
if you have more specific Q please let me /us know
Ash
On Thu, Sep 29, 2011 at 12:39 PM, Ray jonha...@yahoo.com wrote:
what option or command do i have to take out/uncheck inorder to see the
q931 cause code id 8081 in the CUCM traces.. because i set it up and
confirmed all the commands in the workbook and everything is working fine..i
can't generate the cause code 8081..Anyone knows how to create the q931
cause code id 8081?
From: Ray jonha...@yahoo.com
To: ccie voice ccie_voice@onlinestudylist.com
Sent: Thursday, September 29, 2011 10:37 AM
Subject: [OSL | CCIE_Voice] lab3 gatekeeper
what option or command do i have to take out inorder to the the q931 cause
code id 8081.. because i set it up and confired all the commands in the
workbook and everything is working fine..i can't generate the cause code
8081..
HQ#
gatkeeper
zone local GK ccievoice.com 142.1.64.254
zone remote BBGK cisco.com 155.26.1.100 1719
zone prefix BBGK 01144*
zone prefix BBGK 44*
no shut
___
For more information regarding industry leading CCIE Lab training, please
visit www.ipexpert.com
Are you a CCNP or CCIE and looking for a job? Check out
www.PlatinumPlacement.com
___
For more information regarding industry leading CCIE Lab training, please
visit www.ipexpert.com
Are you a CCNP or CCIE and looking for a job? Check out
www.PlatinumPlacement.com
___
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com
Are you a CCNP or CCIE and looking for a job? Check out
www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] IOS software MTP for g711alaw
the mtp will have by default G711u once you add it and as the mtp accept on codec it was complaining , you need to do no codec and then add whatever codec you want , and Yes it will accept g711a Thanks Ash On Wed, Sep 28, 2011 at 8:10 AM, Gerence Guan cisco.g...@gmail.com wrote: Hi Everyone, Is that possible to configure software MTP for g711alaw on the router? I got the following configure from a 2911 router in production: dspfarm profile 4 mtp codec g711alaw maximum sessions software 100 associate application SCCP but with the new router I've got now, every time when I try to type the command codec g711alaw. it prompt me Codec is already configured for the profile, it is not compatible with codec being configured for MTP service. and the codec g711ulaw will be automatically configured. anyone know why? Regards, Gerence ___ 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] TCS capabilities
Hey All , the TCP (by default) will happen after the connect and it will be using the H245 portion of the H323 family so the command you guys referring to will not show you what is going on and even h245 asn1 will also will not show you anything because what happened is that the CCM is waiting for the far end (which the CUBE ) to send his capability via TCS and the CUBE is also waiting for the CCM to send capability ( the cube will not send his capability either) so the Media capability process inside the H323 protocl will timeout after 10 sec (by default ) so to see the time out happens you can use deb cch323 h245 which i don't recommend because its pretty cryptic and you will not see a clear state about the time out , however in the CCM SDL traces (detailed level) you will see the process timing out clearly and it give you the reason of the timing : 001745916| 2011/05/19 19:18:28.462| 001| SdlSig| H245SessionEstablishedFailure | waitForCapabilitiesExchange | H245Interface(1,100,148,8) | H245SessionManager(1,100,23,8) | (0,0,0,0).0-(*:*) | [R:NP - HP: 0, NP: 5, LP: 1, VLP: 0, LZP: 0 DBP: 0] Thanks Ash ___ 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] MOH with H323 gateway
Hello All , in the explination you gave about the problem you must know what type of moh you got on the HQ and you can confirm that by hear the moh and you will know if its MMOH or unicast and this is very important to figure out because this will turn the troubleshooting to diff direction , also if you have connected call betweeen HQ and BR1 Phone and BR1 phone place HQ on hold what you will get ? in regard of the Multicast config changes , i dont see any reason for that , Note that he is sourcing MOH from the FLASH which mean we are not doing any MMOH routing here , we are flooding the MOH to the specified route under the telephone or call-manager fallback so in this case we dont need any kind of Multicasting setup from the provided info i cannot go anywhere in the troubelshoting , you have to send more detailed info about this issue Thanks Ash On Wed, Sep 28, 2011 at 5:53 AM, DeShon Crayton dcrayto...@comcast.netwrote: Also, confirm that the UCM moh server is using multicast address 239.1.1.1 and incrementing on ip address. ** ** *From:* DeShon Crayton [mailto:dcrayto...@comcast.net] *Sent:* Wednesday, September 28, 2011 8:51 AM *To:* 'Vega Wong'; 'ccie_voice@onlinestudylist.com'; 'whl...@gmail.com' *Subject:* RE: [OSL | CCIE_Voice] MOH with H323 gateway ** ** Hello Vega, ** ** I would add the following: ** ** Config t no ip igmp snopping ** ** int l0 ip pim dense-mode ** ** int fa 0/20 ip pim dense-mode ** ** Confirm that “MOH_CL.wav” is in flash Confirm that “MOH_CL.wav” is properly formatted to be used by the cisco router. Try using the default “music-on-hold.au” that comes with CME for testing purposes. Reboot the router.. ** ** ** ** *From:* Vega Wong [mailto:vega2...@yahoo.com.au] *Sent:* Wednesday, September 28, 2011 7:29 AM *To:* ccie_voice@onlinestudylist.com; DeShon Crayton; whl...@gmail.com *Subject:* Re: [OSL | CCIE_Voice] MOH with H323 gateway ** ** Hi guys I have attached more info for this, hope you can help: ! H323 gw config ! hostname HQ-RTR ! network-clock-participate slot 1 ! dot11 syslog no ip source-route ! ! ip cef ! ! no ip domain lookup ip multicast-routing no ipv6 cef ! multilink bundle-name authenticated ! ! isdn switch-type primary-ni ! ! ! voice-card 0 ! voice-card 1 ! ! ! controller T1 1/0/0 pri-group timeslots 1-3,24 ! controller T1 1/0/1 ! ! ! ! ! interface Loopback0 ip address 10.10.110.1 255.255.255.255 ip pim sparse-dense-mode ! interface GigabitEthernet0/0 no ip address duplex auto speed auto ! interface GigabitEthernet0/0.10 encapsulation dot1Q 10 ip address 10.10.100.1 255.255.255.0 ! interface GigabitEthernet0/0.20 encapsulation dot1Q 20 ip address 10.10.200.3 255.255.255.0 ip helper-address 10.10.210.11 ip pim sparse-dense-mode h323-gateway voip bind srcaddr 10.10.200.3 ! interface GigabitEthernet0/0.30 encapsulation dot1Q 30 ip address 10.10.210.1 255.255.255.0 ! interface GigabitEthernet0/1 no ip address shutdown duplex auto speed auto ! interface Serial0/2/0 no ip address encapsulation frame-relay frame-relay lmi-type ansi ! interface Serial0/2/0.1 point-to-point ip address 10.10.111.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 201 ! interface Serial0/2/0.2 point-to-point ip address 10.10.112.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 202 ! interface Serial0/2/1 no ip address shutdown clock rate 200 ! interface Serial1/0/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice no cdp enable ! router ospf 1 router-id 10.10.100.1 log-adjacency-changes network 10.10.0.0 0.0.255.255 area 0 ! ip forward-protocol nd no ip http server no ip http secure-server ! ! ! control-plane ! ! voice-port 1/0/0:23 ! ccm-manager music-on-hold ! ! dial-peer voice 1 pots incoming called-number . direct-inward-dial ! dial-peer voice 2 voip incoming called-number . codec g711ulaw ! dial-peer voice 911 pots destination-pattern 911 port 1/0/0:23 forward-digits all ! dial-peer voice 5000 voip destination-pattern 212394 session target ipv4:10.10.210.11 codec g711ulaw ! dial-peer voice 5001 voip preference 1 destination-pattern 212394 session target ipv4:10.10.210.10 ! ! ! ! gatekeeper shutdown ! ! telephony-service max-ephones 1 max-dn 1 ip source-address 10.10.200.3 port 2000 max-conferences 8 gain -6 moh MOH_CL.wav multicast moh 239.1.1.1 port 16384 route 10.10.200.3 10.10.110.1 transfer-system full-consult create cnf-files version-stamp 7960 Sep 27 2011 22:59:02 ! -- debug ephone moh EPHONE music-on-hold debugging is enabled HQ-RTR# *Sep 28 11:09:56.658: MoH route If
Re: [OSL | CCIE_Voice] Unity Connection - CUCM Integration
do you have the check box of reconnect to the higher ccm checked in the server config page on unity ? can you make sure that you have the port registered with the sub when it will back ? Ash On Mon, Sep 26, 2011 at 11:39 PM, Ken Wyan kew...@gmail.com wrote: Hi, I have a strange issue of CUCM redundancy with Unity Connection Integration. Device pool (for phones voice mail) has SUB first PUB second. In Unity connection GUI I added SUB first PUB second for AXL servers CUCM servers. For TFTP servers in Unity I added PUB only. (as mentioned in IPX proc guide) Voice Mail ports always register with PUB. ( although their DP has SUB first) Hunt list used for VMail has CUCM group SUB-PUB (in this order). It always shows as unregistered. If I remove SUB from the group then only it shows as registered. If I shutdown SUB ; then voicemail is working. (because all phones , vm ports , vm hunt pilot all registered with PUB only). When SUB comes back voicemail not working. DB Replication between PUB SUB shows ok (code 2) What can I do to get this running with dual CUCM? Ken On Mon, Sep 26, 2011 at 12:29 PM, Ken Wyan kew...@gmail.com wrote: I have Unity Connection to CUCM SCCP integration. Everything works fine. But only following test fails. From Unity Connection Administration --- Telephony Integrations --- Phone System - Port Group --- Servers. I added both Cisco Unified Communications Manager Servers both TFTP Servers. When I click on *ping* for each server gives result : Response Time :: Timed Out But all CCIE Lab tasks work fine. Regards, Wyan ___ 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] mask on uccx port, plus sign +, for AAR?
set the mask on the cucm side , and it will take effect , Ash On Mon, Sep 26, 2011 at 9:18 AM, zamuel del Toro sdelto...@hotmail.com wrote: the uccx port network mask can't be assigned using + sign (ej. +1212394) from uccx admin page, should be assigned manually from cucm admin page even when this unsyncronize the uccx integration?. or do i have to consider adding translation without + sign. that because the route pattern are using +1212 on particion AAR thanks and 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 ___ 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] BR2 dial-peer 9........ overlape 900T, and can't make international calls
the first option and you can vary it by also adding : 9.[1-9].. Ash On Mon, Sep 26, 2011 at 9:22 AM, zamuel del Toro sdelto...@hotmail.com wrote: if the requirement is for national 9 plus 8 any digits, it overlap for 900T and can't make international calls is correct using 9[1-9].. or 9.T even when interdigit timeout? thanks and 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 ___ 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] URGENT ## music on hold issue ##
when you but the caller on hold from the phone , that will use what is in your phone confg page of mrgl and moh user/network resources but when you did the same from the ARC now you are using the setting on the CTI port of the ARC , can you please go ahead and stream UNICAST MOH through the CTI ports of the ARC and check the behavior ? tone on hold is generated when the ccm failed to allocate moh resource of when there is codec mismatch ..double check the cti ports/rp settings once again Ash On Mon, Sep 26, 2011 at 7:42 AM, Gerence Guan cisco.g...@gmail.com wrote: Hi Guys, Working on a customer issue: CUCM 6.1 3825 router with IOS12.4(13) ARC call connect 4.1 Phone_M in location_M, region_M, devicepool_M, MRGL_M(MoH_2 Unicast ) Phone_S(ARC console user) in location_S, region_S, devicepool_S, MRGL_S(MoH_3 Muticast, actually from local gateway) H323Gateway_S in location_S, region_S, devicepool_S, MRGL_S(Muticast MoH, actually from local gateway) with E1 for PSTN All ARC CTI ports and CTI route points in location_S, region_S, devicepool_S, MRGL_S(Muticast MoH, actually from local gateway) All MoH servers in Region_MoH MoH stream on G.711 only (for both CUCM stream and MoH file on Gateway_S) G.729 for inter-region G.711 for intra-region Region_MoH use G.711 to all other regions ## Issue 1 When Phone_M call Phone_S (Phone_S's DN, not ARC queue number) If Phone_S put Phone_M on hold by pressing the hold softkey, Phone_M can hear the Music on Hold If Phone_S put Phone_M on hold from the ARC console, Phone_M hear the Tone on Hold If Phone_M put Phone_S on hold by pressing the hold softkey, Phone_S can hear the Music on Hold as I understand, no matter what device be used to put Phone_M on hold, Phone_M should always get the unicast stream from MoH_2 in G.711. I don't understand why I got different result. ## Issue 2 When Mobild call Phone_S (Phone_S's DN, not ARC queue number) If Phone_S put Mobild on hold by pressing the hold softkey, Mobild can hear the Music on Hold If Phone_S put Mobild on hold from the ARC console, Mobild hear the Tone on Hold same as above, because the H323Gateway_S is in region_S, no matter what device to be used to put the mobile on hold, the mobile should always get the multicast stream from the local MoH file on the multicast IP address configured on MoH_3 in G.711 Anyone can think about any problem I may not noticed? Thanks Best Regards, Gerence ___ 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] background image not working..
remove the / from here your command tftp-server flash:/Desktops/320x212x12/List.xml the correct command : tftp-server flash:Desktops/320x212x12/List.xml Ash On Thu, Sep 22, 2011 at 8:40 AM, Ray jonha...@yahoo.com wrote: has anyone done background image and it works.. when i go to user preferencbackgroundimage it says selection unavailable. help! i done know if it is my image.. or what. if anyone has an image that works please forward to me.. I created my in windows 7 paint program and save it as png file. br2# br2#more flash:/Desktops/320x212x12/List.xml CiscoIPPhoneImageList ImageItem Image=TFTP:Desktops/320x212x12/TN-voiceslarge.png URL=TFTP:Desktops/320x212x12/voicesmall.png/ /CiscoIPPhoneImageList br2# tftp-server flash:/Desktops/320x212x12/List.xml tftp-server flash:flash:/Desktops/320x212x12/voicesmall.png tftp-server flash:flash:/Desktops/320x212x12/TN-voiceslarge.png br2#sho flash 0 Nov 17 2009 22:44:36 Desktops/320x212x12 30 131470 Nov 17 2009 22:44:38 Desktops/320x212x12/CampusNight.png 31 80565 Nov 17 2009 22:44:40 Desktops/320x212x12/CiscoFountain.png 32 8156 Nov 17 2009 22:44:40 Desktops/320x212x12/CiscoLogo.png 33 138278 Nov 17 2009 22:44:40 Desktops/320x212x12/Fountain.png 34 165 Sep 21 2011 15:17:18 Desktops/320x212x12/List.xml 35 109076 Nov 17 2009 22:44:42 Desktops/320x212x12/MorroRock.png 36 108087 Nov 17 2009 22:44:44 Desktops/320x212x12/NantucketFlowers.png 37 10820 Nov 17 2009 22:44:44 Desktops/320x212x12/TN-CampusNight.png 38 9657 Nov 17 2009 22:44:46 Desktops/320x212x12/TN-CiscoFountain.png 39 2089 Nov 17 2009 22:44:46 Desktops/320x212x12/TN-CiscoLogo.png 40 7953 Nov 17 2009 22:44:46 Desktops/320x212x12/TN-Fountain.png 41 7274 Nov 17 2009 22:44:46 Desktops/320x212x12/TN-MorroRock.png 42 9933 Nov 17 2009 22:44:48 Desktops/320x212x12/TN-NantucketFlowers.png 43 4651 Sep 21 2011 15:15:04 Desktops/320x212x12/TN-voiceslarge.png 44 3961 Sep 21 2011 15:15:54 Desktops/320x212x12/voicesmall.png ___ 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] Problem with CUCM and CUC SIP integration
that might be a call routing matte , maybe the call from that phones didnt reached the VM pilot at all (css , pt , Tanslation etc,,) ofcourse we cannot be accurate in the answer because its your dial-plan but you can do digit analyizer for that call and see what you will get , if all is fine , ccm sdi traces will tell us what is going on Ash On Tue, Sep 20, 2011 at 12:00 AM, Rynard Coetzee rynard.coet...@bytes.co.za wrote: Hi I am trying to get the SIP integration between CUC and CUCM to work ,but I am stuck at the moment . From my HQ and Branch 1 phones I can dial the UC pilot and have UC answer ,I can also sign into the mailboxes. Problem is when I call from any other phone ,I get a fastbusy as soon as the call gets forwarded to UC ,I have used the Unity Port status monitor but it seems that the call does not get to UC and that it is failing somewhere on the CUCM. Any ideas ? Regards Rynard ___ 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] Problem with CUCM and CUC SIP integration
the ccm language will not affect the traces , you just bring them and attach them i will take a look and share what i will find Ash On Tue, Sep 20, 2011 at 4:34 AM, Rynard Coetzee rynard.coet...@bytes.co.za wrote: I have done DNA and the call is being routed to the SIP trunk correctly ,also the VM pilot points to a Route Pattern which in turn points to the SIP trunk to CUC. I know the trunk is working fine because the phones can call the VM pilot direct. I am busy pulling the sdi traces ,but those traces might as well be in French or something as I have no idea how to decipher them . -Original Message- From: Ashraf Ayyash [mailto:ash.ayy...@gmail.com] Sent: 20 September 2011 01:24 PM To: Rynard Coetzee Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Problem with CUCM and CUC SIP integration that might be a call routing matte , maybe the call from that phones didnt reached the VM pilot at all (css , pt , Tanslation etc,,) ofcourse we cannot be accurate in the answer because its your dial-plan but you can do digit analyizer for that call and see what you will get , if all is fine , ccm sdi traces will tell us what is going on Ash On Tue, Sep 20, 2011 at 12:00 AM, Rynard Coetzee rynard.coet...@bytes.co.za wrote: Hi I am trying to get the SIP integration between CUC and CUCM to work ,but I am stuck at the moment . From my HQ and Branch 1 phones I can dial the UC pilot and have UC answer ,I can also sign into the mailboxes. Problem is when I call from any other phone ,I get a fastbusy as soon as the call gets forwarded to UC ,I have used the Unity Port status monitor but it seems that the call does not get to UC and that it is failing somewhere on the CUCM. Any ideas ? Regards Rynard ___ 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] cBarge problem using MVA
Hey Rynard , as you are using Cbarge , there is no need for Built in Bridge , you will need have the privacy off and the barge method as Cbarge from the phones pages , reset the phone and check the behavior ( make sure that you will have conf brigde available for the phones inside your MRGL Ash On Mon, Sep 19, 2011 at 1:02 AM, Rynard Coetzee rynard.coet...@bytes.co.za wrote: Hi I have issue to get cBarge to work when dialling in via my MVA number. When I dial the MVA number and then start a new call to my HQ phone ,my Branch 2 shows the Remote in Use ,but when I try to Cbarge into the call ,I just get a fastbusy. I have Hardware conference bridge setup for both HQ and BR2 phones ,any ideas what else the problem might be ? Also Built-in bridge and Privacy have been set on both phones. ___ 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] Resolved GK Calls disconnect after hold
Nice one , thank you for sharing this issue Ash On Mon, Sep 19, 2011 at 2:38 AM, DeShon Crayton dcrayto...@comcast.net wrote: Thanks for the input. The fix that was outlined resolved the issue. For notes the scenario was as follows: 1. Use GK to call PSTN via g729 codec 2. Only use g711 moh My issue was that once I took a GK pstn call off of hold, the call would then disconnect. I used debug voip ccapi and debug h245 asn1 to trouble shoot. The following was need to get the proper functionality: MOH server 1. MOH Server is tied to a g711 only Region/Device Pool 2. MOH server needed a MRGL that included a transcoder GK Trunk 1. GK Trunk is tied to a g729 only Region/Device Pool 2. MTP Required needed to be checked 3. GK MRGL included a. MOH server/servers b. g729 MTP resource ( I used a g729 software MTP) I. The g729 MTP is registered to UCM Thanks -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of DeShon Crayton Sent: Sunday, September 18, 2011 8:11 PM To: 'Ashraf Ayyash' Cc: 'OSL Voice' Subject: Re: [OSL | CCIE_Voice] GK Calls disconnect after hold Thanks, I will give the fix a try.. -Original Message- From: Ashraf Ayyash [mailto:ash.ayy...@gmail.com] Sent: Sunday, September 18, 2011 7:15 PM To: DeShon Crayton Cc: OSL Voice Subject: Re: [OSL | CCIE_Voice] GK Calls disconnect after hold forgot to mention , the key to fix this issue (or to isolate it ) is to know who disconnect the call and where exactly the OLC doesn't worked after the resume at the CUBE or at the CUCM ? in such similar call flow , we have used G729r IOS mtp registered with CCM and enabled FS to get this issue fixed , check this as well Ash On Sun, Sep 18, 2011 at 4:12 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote: can you get what cause the call got when it disconnected ? debug ccapi inout at the GK router , also for such issue CCM SDI/SDL traces is needed can you get them ? that scenario worked for me before , i dont have lab to test it now but i would give it try once again and try to produce the issue you are facing Ash On Sun, Sep 18, 2011 at 1:59 PM, DeShon Crayton dcrayto...@comcast.net wrote: I have an interesting issue.. Gatekeepeer calls to the pstn disconnect after I take them off hold.. The call flow is from a sccp UCM phone to the pstn via gatekeeper. The call connects as designed with 2 way audio. From the pstn phone, hold and resume works without issue. The pstn phone stays on-hold, with music, until I hit the resume softkey. Two way audio is not restored after the session is held, then the call disconnects about 10 seconds later. I have the following setup: UCM 7.0.1.11000-2 SCCP firmware 8.4.1S IOS 12.4.(15)T14 The GK trunk is in a g729 only region. The MRGL attached to the trunk has the following resources: 1. sub unicast moh 2. pub unicast moh 3. rsvp software mtp resource (calls to br2) Require MTP nor Wait for H.245 TCS is not checked on the UCM GK Trunk. The moh servers are in a g711 only region, but I set UCM to use g729 between the g711/g729 regions. The moh servers have transcoders available via their device pool setting. Telephony-services is setup on the ios device. The resources attached to the ios device are: 1. transcoding 2. g729 software mtp voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip h323 interface Loopback0 ip address 10.10.1.1 255.255.255.255 h323-gateway voip interface h323-gateway voip id HQRTR ipaddr 10.10.1.1 1719 h323-gateway voip h323-id hq-rtr h323-gateway voip bind srcaddr 10.10.1.1 sccp ccm group 2 bind interface Loopback0 associate ccm 3 priority 1 associate profile 11 register gk-mtp associate profile 10 register gk-xcoder dspfarm profile 10 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 4 associate application SCCP dspfarm profile 11 mtp codec g729r8 maximum sessions software 2 associate application SCCP dial-peer voice 2300 voip destination-pattern 23+ session target ras incoming called-number . dtmf-relay h245-alphanumeric ip qos dscp cs3 signaling no vad gateway ! ! gatekeeper zone local HQRTR cisco.com 10.10.1.1 zone local GK cisco.om zone remote PSTN-WAN pstn.com 10.10.4.1 1719 outvia HQRTR zone prefix GK 1... gw-priority 10 ucm_2 zone prefix GK 1... gw-priority 9 ucm_1 zone prefix GK 1... gw-priority 0 br2-rtr zone prefix PSTN-WAN 23* zone prefix GK 3... gw-priority 10 br2-rtr zone prefix GK 3... gw-priority 0 ucm_1 ucm_2 zone prefix GK 5... gw-priority 9 ucm_1 ucm_2 zone prefix GK 5
Re: [OSL | CCIE_Voice] cBarge problem using MVA
nice , to add to this , when the CCM setup conf call , he will ask for new call BW so take this to the consideration for the future if you will run on any similar issue Ash On Mon, Sep 19, 2011 at 5:29 AM, Rynard Coetzee rynard.coet...@bytes.co.za wrote: Hi Ash Thanks for the reply ,I turned off the built-in bridge but was still getting the error ,I then realised I had Locations CAC set from previous part of the lab and that this was causing the cbarge to fail. Working now. Thanks -Original Message- From: Ashraf Ayyash [mailto:ash.ayy...@gmail.com] Sent: 19 September 2011 02:25 PM To: Rynard Coetzee Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] cBarge problem using MVA Hey Rynard , as you are using Cbarge , there is no need for Built in Bridge , you will need have the privacy off and the barge method as Cbarge from the phones pages , reset the phone and check the behavior ( make sure that you will have conf brigde available for the phones inside your MRGL Ash On Mon, Sep 19, 2011 at 1:02 AM, Rynard Coetzee rynard.coet...@bytes.co.za wrote: Hi I have issue to get cBarge to work when dialling in via my MVA number. When I dial the MVA number and then start a new call to my HQ phone ,my Branch 2 shows the Remote in Use ,but when I try to Cbarge into the call ,I just get a fastbusy. I have Hardware conference bridge setup for both HQ and BR2 phones ,any ideas what else the problem might be ? Also Built-in bridge and Privacy have been set on both phones. ___ 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] ntp
Hello All , i have review the NTP principle again with my Colleagues and with Amit who did lab for this ( Thanks ALOT) and we have done some show commands on the CCM and the IOS in regard of this issue what we found is that the the lowest stratum is the Best and stratum -0 is the lowest value we have in Cisco world, what Justin and the other stated is correct , we will not need Master command to sync the other internal network components to the HQ router , its enough to sync HQ with the UTC based server and then the other routers should take the time correctly from the HQ on UTC based as well the CCM pub is stratum 11 by default and the SUB is 12 , do utils ntp status before you sync them , so based on the above , to sync the CCM with the HQ ntp please make sure that you have Stratum lower than 11 at your HQ and you should be good to go , Justain , Amit , everyone , thank you for sharing / discussing this issue , this is very helpful and add new info for me and i hope the same happened for another people Best Regards Ash On Sun, Sep 18, 2011 at 11:11 PM, Justin Barksdale jus...@barksdale.net wrote: If this is the case then the address you are pointing to at the pstn is incorrect. If you add the ntp server to the router and do show ntp status the router MUST be syncing with an external source or UCM will not sync. Furthermore a reboot is not required on UCM or ther router. Simply add the address to UCM and do utils ntp restart from cli. The key is that the ntp source the router is syncing with must be valid. Sent from my iPhone 4. On Sep 19, 2011, at 1:12 AM, Ray jonha...@yahoo.com wrote: Ntp is not slow when i put the ntp master command on HQ router and the cucm syn right the way. i took off the ntp master from the hq and only have ntp server and ntp source lo0. and reboot the cucm and the hq. and let it set for 4hrs to 5hrs and the cucm would not syn, but the moment i put the ntp master it syns within 2 mins... you can try it in you lab and let me know.. make sure you reboot the cucm and the hq router... From: Justin Barksdale jus...@barksdale.net To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Sunday, September 18, 2011 9:14 PM Subject: Re: [OSL | CCIE_Voice] ntp NTP is by nature slow. An earlier post pointed out the correct method. If the question wants you to sync hq with the pstn and then ccm with hq then the ntp master command will break this requirement. All that is required on the HQ router is ntp server x.x.x.x This is also mentioned in Vic's class class as well as Cisco documentation. The important key here is that the ntp master command can cause you to override a valid external time source and thus can be very dangerous. As long as the ntp server command points to a valid and reachable ntp source then the hq router will sync and you WILL be able to sync ucm without the ntp master command. Justin Barksdale CCIE# 29866 Voice Sent from my iPhone 4. Please excuse any typos. On Sep 18, 2011, at 7:39 PM, ccie_voice-requ...@onlinestudylist.com wrote: Re: [OSL | CCIE_Voice] ntp ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] ntp
Hi all , if we took off the master command we will not be able to sync our internal network entities to the HQ router , please feel free to correct me if i am wrong , the output you gave Ray for the first email and the config say that you have added the master command before you synced with the external ntp and so your internal router got the highest level , to get rid of this you can configure the ntp server and then make sure that you are synced and then use the master command with lower *stratum level and this should do the trick for you , below the link : http://www.cisco.com/en/US/docs/ios/12_1/configfun/configuration/guide/fcd303.html#wp1004877 Ash * On Sat, Sep 17, 2011 at 11:35 PM, Ray jonha...@yahoo.com wrote: i found the issue , i took out the ntp master and then shut down/ no shut my f0/0 connecting to the UTC server, and the clock was syncing as below.. good hq#sh ntp ass address ref clock st when poll reach delay offset disp *~157.26.1.100127.127.1.1 15 63 64 377 0.000 2901006 3.641 * sys.peer, # selected, + candidate, - outlyer, x falseticker, ~ configured hq#sho run | s ntp ntp source Loopback0 ntp server 157.26.1.100 --- On *Sat, 9/17/11, Marko Milivojevic mar...@ipexpert.com* wrote: From: Marko Milivojevic mar...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] ntp To: Ray jonha...@yahoo.com Cc: ccie voice ccie_voice@onlinestudylist.com Date: Saturday, September 17, 2011, 11:44 PM Why did you configure your router to be NTP master? When you did that, on which stratum did your router operate? If it was less than 15, will it sync to a server on stratum 15? Answer these questions and you'll know the answer to yours :-) -- Marko Milivojevic - CCIE #18427 Senior Technical Instructor - IPexpert FREE CCIE training: http://bit.ly/vLecture Mailto: mar...@ipexpert.com http://mc/compose?to=mar...@ipexpert.com Telephone: +1.810.326.1444 Web: http://www.ipexpert.com/ On Sat, Sep 17, 2011 at 18:14, Ray jonha...@yahoo.comhttp://mc/compose?to=jonha...@yahoo.com wrote: looking at the sho ntp ass below and the config below. I could not make the Hq router syn its time from 1567.26.1.100. this question troubled me when i took the exams. any idea!!!.. the UTC server at 157.26.1.100 was set to stratum 15 i think... so how can u make HQ syn time from the UTC server... I was confused here.. ntp source Loopback0 ntp master ntp server 157.26.1.100 hq#sho ntp ass address ref clock st when poll reach delay offset disp *~127.127.1.1 .LOCL. 7 12 16 377 0.000 0.000 0.238 ~157.26.1.10078.85.76.76 16 34 64 376 0.000 2901016 2.591 * sys.peer, # selected, + candidate, - outlyer, x falseticker, ~ configured ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
