Re: [OSL | CCIE_Voice] (no subject)
check also did the mva was enable for the user under end user page. On Sat, Aug 27, 2011 at 1:00 AM, Mini Me cciev.min...@gmail.com wrote: Did you enable it in Service Parameters? HTH From: Ray jonha...@yahoo.com Reply-To: Ray jonha...@yahoo.com Date: Fri, 26 Aug 2011 08:32:19 -0700 (PDT) To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] (no subject) when i dial 3033300 from 525 pstn 2nd and put the pin 12345, MVA IVR ask me to press 2 to turn on the remote dest or press 3 to turn it off.. it doesnt let me dial any extension.. help SB#sho run Building configuration... Current configuration : 5001 bytes ! ! Last configuration change at 05:10:01 CST Fri Aug 26 2011 ! NVRAM config last updated at 23:36:37 CST Mon Aug 22 2011 ! version 15.0 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname BR1 ! boot-start-marker boot-end-marker ! ! no aaa new-model ! ! ! clock timezone CST -6 clock summer-time CST recurring network-clock-participate wic 2 ! dot11 syslog ip source-route ! ! ip cef ! ip dhcp pool phone network 177.2.11.0 255.255.255.0 default-router 177.2.11.1 option 150 ip 10.11.11.19 ! ! no ipv6 cef ! multilink bundle-name authenticated ! ! ! ! isdn switch-type primary-ni ! ! ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 ! voice class h323 1 h225 timeout tcp establish 3 call start slow ! ! ! voice translation-rule 1 rule 1 // // type any unknown plan any isdn ! voice translation-rule 2 rule 1 // // type any subscriber plan any isdn ! voice translation-rule 3 rule 1 // // type any national plan any isdn ! voice translation-rule 4 rule 1 // // type any international plan any isdn ! voice translation-rule 5 rule 1 /.*\(3...\)/ /\1/ ! voice translation-rule 11 rule 1 /^3...$/ /972303/ type any unknown plan any isdn ! voice translation-rule 12 rule 1 /^3...$/ /303/ type any subscriber plan any isdn ! voice translation-rule 13 rule 1 /^3...$/ /972303/ type any national plan any isdn ! voice translation-rule 14 rule 1 /^3...$/ /+1972303/ type any international plan any isdn rule 2 /^4...$/ /+8522404/ type any international plan any isdn ! ! voice translation-profile 911 translate calling 11 translate called 1 ! voice translation-profile INC translate called 5 ! voice translation-profile intl translate calling 14 translate called 4 ! voice translation-profile local translate calling 12 translate called 2 ! voice translation-profile nat translate calling 13 translate called 3 ! ! voice-card 0 ! ! application service mva http://10.11.11.19:8080/ccmivr/pages/IVRMainpage.vxml ! ! ! ! ! ! license udi pid CISCO2811 sn FTX0902D1ZG ! redundancy ! ! controller T1 0/2/0 pri-group timeslots 1-4,24 ! ! ! ! ! ! ! ! ! interface Loopback0 ip address 18.1.1.1 255.255.255.0 ip ospf network point-to-point h323-gateway voip interface h323-gateway voip bind srcaddr 18.1.1.1 ! ! interface FastEthernet0/0 no ip address shutdown duplex auto speed auto ! ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! ! interface FastEthernet0/1/0 switchport trunk native vlan 12 switchport mode trunk switchport voice vlan 11 spanning-tree portfast ! ! interface FastEthernet0/1/1 switchport trunk native vlan 12 switchport mode trunk switchport voice vlan 11 spanning-tree portfast ! ! interface FastEthernet0/1/2 switchport trunk native vlan 12 switchport mode trunk switchport voice vlan 11 spanning-tree portfast ! ! interface FastEthernet0/1/3 switchport trunk native vlan 12 switchport mode trunk switchport voice vlan 11 spanning-tree portfast ! ! interface Serial0/0/0 no ip address encapsulation frame-relay no frame-relay inverse-arp ! ! interface Serial0/0/0.101 point-to-point ip address 177.0.101.2 255.255.255.0 frame-relay interface-dlci 602 ! interface Serial0/0/1 no ip address shutdown clock rate 200 ! ! interface Serial0/2/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn outgoing ie facility isdn outgoing display-ie no cdp enable ! ! interface Vlan1 no ip address ! ! interface Vlan11 ip address 177.2.11.1 255.255.255.0 ip helper-address 10.11.11.19 ! ! interface Vlan12 ip address 177.2.12.1 255.255.255.0 ! ! ! router ospf 1 log-adjacency-changes network 18.1.1.1 0.0.0.0 area 0 network 177.0.101.0 0.0.0.255 area 0 network 177.2.11.0 0.0.0.255 area 0 network 177.2.12.0 0.0.0.255 area 0 ! ip forward-protocol nd no ip http server no ip http secure-server ! ! ! ! ! ! ! ! ! control-plane ! ! ! voice-port 0/2/0:23 translation-profile incoming INC ! ! mgcp fax t38 ecm !
[OSL | CCIE_Voice] AIM-CUE CF Card Issue
Hi Guys, Hope someone can assist me on the following issue. I've an AIM-CUE module which one day after the power cycle, I got the following error during the module boot up. Not a cisco supported CF. Please use cisco supported CF and reinstall the software. Understand my CF card could be toasted as it can't even detected by the card reader. I've hence doing some google and saw the original AIM-CUE-1GBCF= cost even more than the module itself. then someone mentioned an CF card with 2001888 sectors could be a cheaper replacement. I've success to found two 1GB CF card with the same sector (one of it is cisco router used cf 1gb), and since have a chance to borrow the other working AIM-CUE CF card on hand, I've used fedora core 15 and DDed the working CF to an Image with the following command dd if=/dev/sdb of=/image bs=32768 and write the image to the two card that i've bought, but guess what, i've not gotten any luck on this. Still after using the bootloader to initialize the module, i still getting the not a cisco supported CF error. By any chance anyone successfully to run AIM-CUE on other CF card and can shed some light? P/S I'm using this module for my CCIE lab preparation, and it is a used unit, hence there isn't anyway I could do RMA : ( Regards, Alex ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] voice traffic EF outbound / inbount packet count
Any kind soul able to advise? On Thu, Aug 4, 2011 at 12:20 PM, Alex Goh ncsalex@gmail.com wrote: Hi Guys, Hope I can seek a little help here. I've MLP QOS configured on my lab and I notice that the ef packet count for outbound and inbound on the router WAN interface are almost identical. For example I've two office, HQ and Branch, both office have phones registered to CUCM located at HQ and there is a 384 WAN link between. both voice gateway are running MGCP. When the call is established, I notice the following output: HQ Router Service-policy input: EF-Inbound Class-map: AutoQoS-VoIP-RTP-Trust (match-any) *332* packets, 21248 bytes 5 minute offered rate 6000 bps Match: ip dscp ef (46) 332 packets, 21248 bytes 5 minute rate 6000 bps Class-map: class-default (match-any) 11 packets, 674 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: any Service-policy output: AutoQoS-Policy-Trust queue stats for all priority classes: Queueing queue limit 64 packets (queue depth/total drops/no-buffer drops) 0/0/0 (pkts output/bytes output) 335/20770 Class-map: AutoQoS-VoIP-RTP-Trust (match-any) *335* packets, 20770 bytes 5 minute offered rate 8000 bps, drop rate 0 bps Match: ip dscp ef (46) 335 packets, 20770 bytes 5 minute rate 8000 bps Priority: 33% (126 kbps), burst bytes 3150, b/w exceed drops: 0 Branch Router Virtual-Access3 Service-policy input: EF-Inbound Class-map: AutoQoS-VoIP-RTP-UnTrust (match-any) *190* packets, 12160 bytes 5 minute offered rate 1 bps Match: protocol rtp audio 190 packets, 12160 bytes 5 minute rate 1 bps Match: access-group name AutoQoS-VoIP-RTCP 0 packets, 0 bytes 5 minute rate 0 bps Class-map: class-default (match-any) 12 packets, 1712 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: any Service-policy output: AutoQoS-Policy-UnTrust queue stats for all priority classes: Queueing queue limit 64 packets (queue depth/total drops/no-buffer drops) 0/0/0 (pkts output/bytes output) 188/11656 Class-map: AutoQoS-VoIP-RTP-UnTrust (match-any) *187* packets, 11594 bytes 5 minute offered rate 1 bps, drop rate 0 bps Match: protocol rtp audio 187 packets, 11594 bytes 5 minute rate 1 bps Match: access-group name AutoQoS-VoIP-RTCP 0 packets, 0 bytes 5 minute rate 0 bps QoS Set dscp ef Packets marked 188 Priority: 33% (126 kbps), burst bytes 3150, b/w exceed drops: 0 As my understanding, aren't the inbound ef packet count should be same or at least close to the number of the sender's outbound? for this example HQ is sending 335 packet and isn't Branch should be receiving 335 ef packet? (assuming no packet lost) Also i notice on the same router, the inbound / outbound ef packet count is almost identical, is that correct pattern for voice traffic? When checking the statistics on the phone and is showing the same. if I've have huge number mismatch on this will that means the call suffer quality issue? Appreciate if someone can enlighten me on this. Thanks Regards, Alex ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] B-ACD call queue display on 7941
Hi Guys, I was trying the b-acd drop-through mode, and notice that number of call in queue display doesn't show on the my 7941 if the hunt member is line 2 of my 7941. but if i change the line 1 as the hunt member, i can see the queue display. However my other phone 7961 has no such issue when i configure line 1 or line 2 as member. I've check the both phones's firmware and it is the same. i have the following config ephone-dn 2 octo-line number 5003 name BR2 PHONE 3 ephone-dn-template 1 ! ephone-dn 3 octo-line number 5001 name BR2 PHONE 1 ephone-dn-template 1 ephone-dn 9 dual-line number 5101 no-reg both name Agent ! ephone-dn 10 dual-line number 5102 no-reg both name Agent ephone-hunt 1 longest-idle pilot 5100 list 5101, 5102 ephone 2 device-security-mode none mac-address ephone-template 1 type 7941 button 1:2 2:10 ! ephone 3 device-security-mode none mac-address ephone-template 1 type 7961 button 1:3 2:9 anyone encountered this before or is it 7941 phones behavior? Regards, Alex ___ For more information regarding industry leading 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] Presence, CUPS and CUPC
Hi Randall, Thanks for the info. but I notice that this way, the contact is not dialable. the contact's presence status can be seen and IM is working, but not calling the contact using deskphone/softphone mode. when edit the contact's details the phone number is not saved. is that the normal behaviour when LDAP is not used? Regards, Alex On Mon, Jul 25, 2011 at 10:23 PM, Randall Saborio ill2...@gmail.com wrote: That is correct. It is because of not using LDAP. The alternative is to log into http://cups-server/ccmuser and log in with each user, then on the directory or contacts, create the contact from there where it works regardless of ldap integration. On Sat, Jul 23, 2011 at 11:10 PM, Vega Wong vega2...@yahoo.com.au wrote: Hi Thanks for the advices, I sort out the offline issue, it was because I didnt setup DNS for CUPC to connect to CUPS. Once thats setup, IM works as well as Presence status. Just one more thing, by manually adding the contact, I couldnt save the phone number of the contact. That means I cant place a call to the contact in CUPC. But it works when I use the phone control (in CUPC) and actually dial the extension. Is this also due to not using ldap? Cheers -- * From: * Adil Shaikh adil.sha...@gmail.com; * To: * Vega Wong vega2...@yahoo.com.au; * Cc: * ccie_voice@onlinestudylist.com; * Subject: * Re: [OSL | CCIE_Voice] Presence, CUPS and CUPC * Sent: * Sun, Jul 24, 2011 4:51:08 AM Hi Vega, You wrote: However, at the bottom of the CUPC, it always shown as Connected(limited). Also, I cant search the contact within the CUPS.With the contact added, it always shown as offline. Connected (Limited) seems to be normal behaviour in absense of LDAP. If you have added contact from the User page in CUPS then it should be working provided in CUCM you have done Line Association with the User. (that is on DN page, select the device, press line Edit Line Appearance then at the bottom of the page Associate End User). HTH -adil On Fri, Jul 22, 2011 at 11:25 PM, Vega Wong vega2...@yahoo.com.auwrote: Hi Experts I am working on CUPS and CUPC at the moment, I just have some questions with the setup. So far I have done: - I have successfully set up the integration between CUCM and CUPS. As I can see all the users and the SIP trunks automatically appears in CUPS. - IPPM works on IP phones, I can send messages between IP Phones - I can add contacts using IPPM, or the User page of CUPS. The contacts will shows up in CUPC - When I run the system troubleshooter in CUPS, no Red crosses shown. (Except those items I havent configured - LDAP, voicemail) My issue is with the CUPC, I can log in the system using the User ID. I can see the contact added through IPPM or CUPS User page. I can use the CUPC to control the IP phone. However, at the bottom of the CUPC, it always shown as Connected(limited). Also, I cant search the contact within the CUPS.With the contact added, it always shown as offline. I have read that I will need LDAP in order to make the presence status to work in CUPC, is that true? Can i make this work without the LDAP? Please help Cheers ___ For more information regarding industry leading 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 -- Randall da ill Saborio CCIE Voice Wannabe #10054675811 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] MLP on FR issue
Hi Guys, Hope I can seek a little help here, I've encountered the MLP broke the WAN link issue, which can't be resolved by any work around discussed before. What happened is I'm trying to configured an MLP for HQ and BR1 with auto qos and also manually, but both landed me on broken WAN where router reboot on HQ and BR1, even PSTN doesn't help. I've also try to remove the frame-relay interface-dlci command and add it back but no luck. The virtual template never having a status UP and no connectivity between HQ and BR1 Appreciate if any can help on this. Regards, Alex My config is as below: *HQ:* class-map match-any AutoQoS-VoIP-RTP-Trust match ip dscp ef class-map match-any AutoQoS-VoIP-Control-Trust match ip dscp cs3 match ip dscp af31 ! ! policy-map AutoQoS-Policy-Trust class AutoQoS-VoIP-RTP-Trust priority percent 33 class AutoQoS-VoIP-Control-Trust bandwidth percent 5 class class-default fair-queue interface Serial0/0/1:0 no ip address encapsulation frame-relay frame-relay traffic-shaping interface Serial0/0/1:0.101 point-to-point description WAN Link to BR1 bandwidth 384 snmp trap link-status frame-relay interface-dlci 101 ppp Virtual-Template1 class AutoQoS-FR-Se0/0/1:0-101 interface Virtual-Template1 bandwidth 384 ip address 142.1.67.1 255.255.255.252 ppp multilink ppp multilink interleave ppp multilink fragment delay 10 service-policy output AutoQoS-Policy-Trust map-class frame-relay AutoQoS-FR-Se0/0/1:0-101 frame-relay cir 364800 frame-relay bc 3648 frame-relay be 0 frame-relay mincir 364800 Interface IP-Address OK? Method Status Protocol Serial0/0/1:0.101 unassigned YES NVRAM up up Serial0/0/1:0.201 142.1.67.5 YES NVRAM up up Virtual-Access1unassigned YES unset down down Virtual-Template1 142.1.67.1 YES manual down down Virtual-Access2142.1.67.1 YES TFTP up up Virtual-Access3142.1.67.1 YES TFTP up up (config-subif)#do sh ppp multilink Virtual-Access3 Bundle name: R2 Remote Endpoint Discriminator: [1] BR1 Local Endpoint Discriminator: [1] HQ Bundle up for 00:14:20, total bandwidth 384, load 1/255 Receive buffer limit 12192 bytes, frag timeout 1000 ms Interleaving enabled 0/0 fragments/bytes in reassembly list 0 lost fragments, 0 reordered 0/0 discarded fragments/bytes, 0 lost received 0x5D received sequence, 0x2 sent sequence Member links: 1 (max not set, min not set) Vi2, since 00:14:20, 480 weight, 470 frag size No inactive multilink interfaces *BR1:* class-map match-any AutoQoS-VoIP-Remark match ip dscp ef match ip dscp cs3 match ip dscp af31 class-map match-any AutoQoS-VoIP-Control-UnTrust match access-group name AutoQoS-VoIP-Control class-map match-any AutoQoS-VoIP-RTP-UnTrust match protocol rtp audio match access-group name AutoQoS-VoIP-RTCP ! ! policy-map AutoQoS-Policy-UnTrust class AutoQoS-VoIP-RTP-UnTrust set dscp ef priority percent 33 class AutoQoS-VoIP-Control-UnTrust bandwidth percent 5 set dscp af31 class AutoQoS-VoIP-Remark set dscp default class class-default fair-queue interface Serial0/0/1:0 no ip address encapsulation frame-relay frame-relay traffic-shaping ! interface Serial0/0/1:0.101 point-to-point bandwidth 384 frame-relay interface-dlci 101 ppp Virtual-Template1 class AutoQoS-FR-Se0/0/1:0-101 ! interface Virtual-Template1 bandwidth 384 ip address 142.1.67.2 255.255.255.252 ppp multilink ppp multilink interleave ppp multilink fragment delay 10 service-policy output AutoQoS-Policy-UnTrust ip access-list extended AutoQoS-VoIP-Control permit tcp any any eq 1720 permit tcp any any range 11000 11999 permit udp any any eq 2427 permit tcp any any eq 2428 permit tcp any any range 2000 2002 permit udp any any eq 1719 permit udp any any eq 5060 ip access-list extended AutoQoS-VoIP-RTCP permit udp any any range 16384 32767 ! ! map-class frame-relay AutoQoS-FR-Se0/0/1:0-101 frame-relay cir 364800 frame-relay bc 3648 frame-relay be 0 frame-relay mincir 364800 Interface IP-Address OK? Method Status Protocol Serial0/0/1:0.101 unassigned YES NVRAM up up Virtual-Access1unassigned YES unset down down Virtual-Template1 142.1.67.2 YES manual down down Virtual-Access2142.1.67.2 YES TFTP up up Virtual-Access3142.1.67.2 YES TFTP up up (config-subif)#do sh ppp multi Virtual-Access3 Bundle name: R1 Remote Endpoint Discriminator: [1] HQ Local Endpoint Discriminator: [1] BR1 Bundle up for 00:14:40, total bandwidth 384, load 1/255 Receive buffer limit 12192 bytes, frag timeout 1000 ms Interleaving enabled 0/0 fragments/bytes in reassembly list 0 lost fragments, 0 reordered 0/0 discarded fragments/bytes, 0 lost received 0x2 received sequence, 0x5F sent sequence Member links: 1 (max not set, min not set)
Re: [OSL | CCIE_Voice] MLP on FR issue
Hi Guys, Managed to resolve the issue. is my OSPF config causing it, as I notice pinging the VT IP on both router is success. Regard, Alex On Sun, Jul 17, 2011 at 9:24 PM, Alex Goh ncsalex@gmail.com wrote: Hi Guys, Hope I can seek a little help here, I've encountered the MLP broke the WAN link issue, which can't be resolved by any work around discussed before. What happened is I'm trying to configured an MLP for HQ and BR1 with auto qos and also manually, but both landed me on broken WAN where router reboot on HQ and BR1, even PSTN doesn't help. I've also try to remove the frame-relay interface-dlci command and add it back but no luck. The virtual template never having a status UP and no connectivity between HQ and BR1 Appreciate if any can help on this. Regards, Alex My config is as below: *HQ:* class-map match-any AutoQoS-VoIP-RTP-Trust match ip dscp ef class-map match-any AutoQoS-VoIP-Control-Trust match ip dscp cs3 match ip dscp af31 ! ! policy-map AutoQoS-Policy-Trust class AutoQoS-VoIP-RTP-Trust priority percent 33 class AutoQoS-VoIP-Control-Trust bandwidth percent 5 class class-default fair-queue interface Serial0/0/1:0 no ip address encapsulation frame-relay frame-relay traffic-shaping interface Serial0/0/1:0.101 point-to-point description WAN Link to BR1 bandwidth 384 snmp trap link-status frame-relay interface-dlci 101 ppp Virtual-Template1 class AutoQoS-FR-Se0/0/1:0-101 interface Virtual-Template1 bandwidth 384 ip address 142.1.67.1 255.255.255.252 ppp multilink ppp multilink interleave ppp multilink fragment delay 10 service-policy output AutoQoS-Policy-Trust map-class frame-relay AutoQoS-FR-Se0/0/1:0-101 frame-relay cir 364800 frame-relay bc 3648 frame-relay be 0 frame-relay mincir 364800 Interface IP-Address OK? Method Status Protocol Serial0/0/1:0.101 unassigned YES NVRAM up up Serial0/0/1:0.201 142.1.67.5 YES NVRAM up up Virtual-Access1unassigned YES unset down down Virtual-Template1 142.1.67.1 YES manual down down Virtual-Access2142.1.67.1 YES TFTP up up Virtual-Access3142.1.67.1 YES TFTP up up (config-subif)#do sh ppp multilink Virtual-Access3 Bundle name: R2 Remote Endpoint Discriminator: [1] BR1 Local Endpoint Discriminator: [1] HQ Bundle up for 00:14:20, total bandwidth 384, load 1/255 Receive buffer limit 12192 bytes, frag timeout 1000 ms Interleaving enabled 0/0 fragments/bytes in reassembly list 0 lost fragments, 0 reordered 0/0 discarded fragments/bytes, 0 lost received 0x5D received sequence, 0x2 sent sequence Member links: 1 (max not set, min not set) Vi2, since 00:14:20, 480 weight, 470 frag size No inactive multilink interfaces *BR1:* class-map match-any AutoQoS-VoIP-Remark match ip dscp ef match ip dscp cs3 match ip dscp af31 class-map match-any AutoQoS-VoIP-Control-UnTrust match access-group name AutoQoS-VoIP-Control class-map match-any AutoQoS-VoIP-RTP-UnTrust match protocol rtp audio match access-group name AutoQoS-VoIP-RTCP ! ! policy-map AutoQoS-Policy-UnTrust class AutoQoS-VoIP-RTP-UnTrust set dscp ef priority percent 33 class AutoQoS-VoIP-Control-UnTrust bandwidth percent 5 set dscp af31 class AutoQoS-VoIP-Remark set dscp default class class-default fair-queue interface Serial0/0/1:0 no ip address encapsulation frame-relay frame-relay traffic-shaping ! interface Serial0/0/1:0.101 point-to-point bandwidth 384 frame-relay interface-dlci 101 ppp Virtual-Template1 class AutoQoS-FR-Se0/0/1:0-101 ! interface Virtual-Template1 bandwidth 384 ip address 142.1.67.2 255.255.255.252 ppp multilink ppp multilink interleave ppp multilink fragment delay 10 service-policy output AutoQoS-Policy-UnTrust ip access-list extended AutoQoS-VoIP-Control permit tcp any any eq 1720 permit tcp any any range 11000 11999 permit udp any any eq 2427 permit tcp any any eq 2428 permit tcp any any range 2000 2002 permit udp any any eq 1719 permit udp any any eq 5060 ip access-list extended AutoQoS-VoIP-RTCP permit udp any any range 16384 32767 ! ! map-class frame-relay AutoQoS-FR-Se0/0/1:0-101 frame-relay cir 364800 frame-relay bc 3648 frame-relay be 0 frame-relay mincir 364800 Interface IP-Address OK? Method Status Protocol Serial0/0/1:0.101 unassigned YES NVRAM up up Virtual-Access1unassigned YES unset down down Virtual-Template1 142.1.67.2 YES manual down down Virtual-Access2142.1.67.2 YES TFTP up up Virtual-Access3142.1.67.2 YES TFTP up up (config-subif)#do sh ppp multi Virtual-Access3 Bundle name: R1 Remote Endpoint Discriminator: [1] HQ Local Endpoint
[OSL | CCIE_Voice] Cisco Live 2011 Voice Techtorial
Hi Guys, By any chance any have the latest CCIE Voice Techtorial from Cisco Live 2011 and willing to share? Thanks, Regards, Alex ___ For more information regarding industry leading 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 Phones on Lab
it would be best it we have access to a copy of the techtorial slides like 2010... On Wed, Jul 13, 2011 at 1:18 AM, Brian Mulgrew btmulg...@hotmail.comwrote: As was already announced, there will not be any Core Knowledge/Open Ended Questions after August 15, 2011 Er.. I thought these were no longer tested as of May 2010? Sent from my iPad On 12 Jul 2011, at 17:38, Chris Martin clm.c...@gmail.com wrote: http://blog.ipexpert.com/2011/07/12/cisco-live-news-and-updates-ccie-voice/#more-7578 http://blog.ipexpert.com/2011/07/12/cisco-live-news-and-updates-ccie-voice/#more-7578 Chris On Tue, Jul 12, 2011 at 10:45 AM, Bill Lake whl...@gmail.com whl...@gmail.com wrote: do you have a link to Jason's blog On Tue, Jul 12, 2011 at 9:57 AM, Chris Martin clm.c...@gmail.com clm.c...@gmail.com wrote: That was really what was said. No SIP endpoints, TODAY, but they could be there within the next two weeks, take that for what you will. Jason's blog entry did cover a lot of the info, I will say there was a lot of emphasis from Ben on troubleshooting, and being able to get info from debugs and traces. IE: they may ask you to post the relevant debug lines for h245 negotiation or SIP negotiations. Chris On Tue, Jul 12, 2011 at 8:59 AM, Bryan Byrne ccie.25...@gmail.com ccie.25...@gmail.com wrote: Jason just posted a blog entry with information from the CCIE Voice session at Cisco Live. In his post he stated There are no SIP phones on the currently available labs. They are looking into including them, but that will require the development of a new lab. Are we to assume that the lab blue print is incorrect and that SIP endpoints for CUCM and CUCME are not something we should be concerned with? Was anyone else in the session that could provide some additional input? -Bryan ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.comwww.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out http://www.PlatinumPlacement.comwww.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.comwww.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out http://www.PlatinumPlacement.comwww.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.comwww.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out http://www.PlatinumPlacement.comwww.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Couldn't browse script/prompt repository
Hi Guys, I'm having a weird issue with my UCCX 7.0 (1) Build 109. When login to the CCX editor, I'm not able to save or browse the script/prompt by clicking the repository button (see attached). I've tried reinstalled my UCCX with the repair options, rebooted my UCCX server, created a new uccx admin, and tried running CCX editor from another box all no lucks. Also, I notice the UCCX only works on default script, whenever I save as a default script like icd.aef without modifying it to the script/system/default folder. the UCCX will turn into partial service state,. where Application Manager is the one that OOS. By the way, the uccxadmin end user account was assigned with CCM Super User and allow CTI control all group, if this is related. Can someone shed some light on this? Regards, Alex attachment: uccxeditor.jpg___ For more information regarding industry leading 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] Couldn't browse script/prompt repository
Hi Ron, Yes I already have an application using the default ICD script, but still I can't browse or save on the editor's repository Hi Santiago, I'm log in with uccxadmin account. I know anonymous account will not be able to run reactive script and browse/save script repository. Regards, Alex On Mon, Jul 4, 2011 at 11:53 PM, Santiago Figueroa sfigue...@mnet.com.mxwrote: ** When you logg in CCX editor to do that uccxadmin? ** ** -- *De:* **ccie_voice-boun...@onlinestudylist.com** [mailto:** ccie_voice-boun...@onlinestudylist.com**] *En nombre de *Alex Goh *Enviado el:* Lunes, 04 de Julio de 2011 09:32 a.m. *Para:* OSL *Asunto:* [OSL | CCIE_Voice] Couldn't browse script/prompt repository ** ** Hi Guys, I'm having a weird issue with my UCCX 7.0 (1) Build 109. When login to the CCX editor, I'm not able to save or browse the script/prompt by clicking the repository button (see attached). I've tried reinstalled my UCCX with the repair options, rebooted my UCCX server, created a new uccx admin, and tried running CCX editor from another box all no lucks. Also, I notice the UCCX only works on default script, whenever I save as a default script like icd.aef without modifying it to the script/system/default folder. the UCCX will turn into partial service state,. where Application Manager is the one that OOS. By the way, the uccxadmin end user account was assigned with CCM Super User and allow CTI control all group, if this is related. Can someone shed some light on this? Regards, Alex -- La información incluida en este mensaje y sus anexos es CONFIDENCIAL y para USO EXCLUSIVO de sus destinatarios. No está permitida su divulgación y/o reproducción sin autorización. Si ha recibido este mensaje y no le incumbe, le rogamos nos los comunique y proceda a su borrado. Gracias. Information included in this e-mail and attached files is CONFIDENTIAL and only for the EXCLUSIVE USE of the receivers. Circulation and/or copy without permission is not allowed. If you have received this e-mail and you are not the intended recipient, please let us know and erase the message and attached files. Thank you. ___ For more information regarding industry leading 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] Passed
Hi George, Big congratulation to you and you deserve it! We appreciate your contribution to the list and hope to see you again. Now go and get back your social life : ) Cheers, Alex On Wed, Jun 29, 2011 at 9:44 PM, George Goglidze gogli...@gmail.com wrote: Hi all, As the subject suggests, it's official, I'm dual CCIE #19926 RS and Voice starting yesterday. Finally, it's a relief, no more studying late, spending weekends on a computer, making calls, my neighbours think I escaped a psychiatric, after hearing voices at night TEST VOICEMAIL FOR HQ PHONE 1 and my personal favourite YOUR POSITION IN QUEUE IS :) I would like to transmit a very special thank you to IPExpert, and especially Vik Malhi. He's definitely made difference for me as a trainer. Just when you think you know it all about something, he would come up with something to prove me wrong, to show me the gaps in my knowledge I didn't know existed. Thank you Vik! Thanks to everyone on this forum too, there are many good people on the forum, with big knowledge, and more importantly willing to share it. It was big fun, I enjoyed the process a lot. By the way, I made it technically on my first attempt. Well, I payed once only, although I went to exam 3 times. 1st time, technical problems, Cisco gave me free retake voucher, 2nd time, again technical problems, again free voucher. To be honest after so many free retakes, I'm not even sure if I really passed, or I was costing them too much so they decided to give it to me :-), good samaritans these cisco guys. Wish you all good luck, and don't get frustrated, and most importantly have fun. 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] FRF.12 question
Hi Duy, Thanks I shall test it out and confirm. my understanding could be wrong. Regards, Alex On Mon, Jun 27, 2011 at 10:04 PM, ccieid1ot ccieid...@gmail.com wrote: Alex, definitely needed. Test it out by enabling it on HQ router than don't enable on BR2 router. Now open up the cue gui and see if it operated correctly. duy ccie #27737 voice tmobile g2 On Jun 27, 2011 6:17 AM, Alex Goh ncsalex@gmail.com wrote: Hi Shrini, I guess what cristobal trying to mean is when he is using class based shaping, instead of FRTS which required the command on physical interface, do he need to care about the qos setting between HQ and BR1. By the way, I have one question though, for the case when FRTS was enable on HQ Physical serial interface, do we need to enable FRTS also on the opposite site? I remember I tried before and WAN link isn't broken... Alex On Mon, Jun 27, 2011 at 3:44 PM, Shrini linuxbos...@gmail.com wrote: ** It looks like not effected but it is. The bandwidth drops to 56k. Good idea is to apply the Br2 service policy to Br1 connected srl interface even you not shaping the traffic. sh frame-relay pvc dlci will provide you the details. Thanks Shrini On 6/26/2011 2:34 PM, Cristobal Priego wrote: hello all when you configure FRF.12 manually on your seial interfaces on HQ and BR2 on the HQ router where the same physical interface is used to connect BR1 and BR2, BR1 link isn't affected at all because traffic shaping isn't enabled on the physical interface, correct ? so i can pretty much ignore that link in regards to a basic QoS config if not needed 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] Setting up home equipment
Hi Randall If I'm not wrong, 7961 and 7941 series support the + dialing from call list, which probably you need it to practice the globalization/localization call routing. Regards, Alex On Wed, Jun 29, 2011 at 8:08 AM, Rrcrumm rrcr...@yahoo.com wrote: Hi I'm setting up a switch, router and phones and the proctorlabs racks. I plan on using 7960 phones because they are cheaper Is there any reason to get 7961's? I'm just keeping the cost down Thanks Randall Sent from my iPhone ___ For more information regarding industry leading 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] GK failover call display
Hi Adil, It is normal or expected behaviour where + sign will not be able to pass to h323 gateway, u will to prefix using translation-rule on h323 gateway or configured incoming prefix on cucm from the CUCM Help page you will see this SIP and MGCP gateways can support sending the international escape character, +, for calls. H.323 gateways do not support the + HTH Regards, Alex On Wed, Jun 29, 2011 at 8:20 AM, Adil Shaikh adil.sha...@gmail.com wrote: Hi all, I have configured route list with 1st choice as gatekeeper and 2nd choice as local PSTN. When I shut down the Gatekeeper, the call goes out from PSTN and back into branch gateway via PSTN as expected. debug isdn q931 shows the 'Calling Party Number' in +E164 format but the phone display calling party number without plus. The phone is 7965. Is this what you are getting on your phone? Is this normal behaviour? One branch site is H323 and other is CME. Thanks -adil -- .. . . _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] IPPM getting invalid device
oso check if you have already assign the user capability under licensing capability On Mon, Jun 27, 2011 at 4:47 AM, Cristobal Priego cristobalpri...@gmail.com wrote: does yourapp user matches the user in CUPS ? is ip phone messenger on in CUPS? do you have the CTI enabled and the CTI controll all devices group associated to your app user ? 2011/6/26 Alan Gardner agard...@ctclc.com Invalid Device You were trying to access IP Phone Messenger service from a device not provisioned on Cisco CallManager server. Please work with your system administrator to get this device configured. I am running CUCM 7 and CUPS 7 and I have completed the following steps:* *** 1. Configured IP PhoneMSG service with CUPS in URL 2. Created PhoneMessenger application user and added IPC and 7965 phones in Controlled Devices 3. Associated end users with primary DNs on IPC and 7965 and configured end users with Standard CTI Enabled user group permissions. 4. Subscribed both phones with IP PhoneMSG service ** ** Any ideas ** ** ** ** Best Regards, ** ** Alan Gardner ** ** ___ For more information regarding industry leading 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] FRF.12 question
Hi Shrini, I guess what cristobal trying to mean is when he is using class based shaping, instead of FRTS which required the command on physical interface, do he need to care about the qos setting between HQ and BR1. By the way, I have one question though, for the case when FRTS was enable on HQ Physical serial interface, do we need to enable FRTS also on the opposite site? I remember I tried before and WAN link isn't broken... Alex On Mon, Jun 27, 2011 at 3:44 PM, Shrini linuxbos...@gmail.com wrote: ** It looks like not effected but it is. The bandwidth drops to 56k. Good idea is to apply the Br2 service policy to Br1 connected srl interface even you not shaping the traffic. sh frame-relay pvc dlci will provide you the details. Thanks Shrini On 6/26/2011 2:34 PM, Cristobal Priego wrote: hello all when you configure FRF.12 manually on your seial interfaces on HQ and BR2 on the HQ router where the same physical interface is used to connect BR1 and BR2, BR1 link isn't affected at all because traffic shaping isn't enabled on the physical interface, correct ? so i can pretty much ignore that link in regards to a basic QoS config if not needed 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] remote dest profile
it should show 5001 On Mon, Jun 27, 2011 at 1:38 PM, donny f f.faraday...@gmail.com wrote: hi, when we config Remote Dest Profile for SNR. When call come from our PSTN (6171234) phone to UCM ext phone , should it show our ext 5001 or showing the SNR (PSTN #) ? for ie :we have SNR6171234and match to 5001. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] how to retrieve list of Active IP Phones ONLY...
Hi Rashid, Looks like there is some issue with the RIS Data Collertor service as per the screen capture. Can you confirm that service is running on all the servers? Regards, Alex On Mon, Jun 27, 2011 at 3:23 PM, Rashid Khan me_rashid...@yahoo.com wrote: Thanks Julien, that resolved my problem but partially not fully. I have 2 clusters here. When I am using this tool for Cluster A, I am getting required results, But when I use it for Cluster B, This tool only showing me 24 Registered Devices even thought there are almost 250 IP Phones register with this cluster. I am also attaching screen shot of RTMT tool output.. Regards Rashid. -- *From:* Julien Krieger krieger.jul...@gmail.com *To:* Rashid Khan me_rashid...@yahoo.com *Cc:* ccie voice ccie_voice@onlinestudylist.com *Sent:* Thu, June 23, 2011 6:37:52 PM *Subject:* Re: [OSL | CCIE_Voice] how to retrieve list of Active IP Phones ONLY... Hi Rashid, RTMT is your tool !!! Download it into the plugin's section Julien 2011/6/23 Rashid Khan me_rashid...@yahoo.com Dear Team, I want to know is there any way to findout a list of Currently active or registed IP Phones with Call Manager. Oneway to do this is, write nothing in Text Box and press Find button, when I do this I also see non active devices Or the devices whose Status is Not found also appearing. I only want list of phones which are Active or working currently, Regards Rashid ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] MOH between HQ and BR1 Phones
Hi Guys, I was trying some MOH question just now and notice something which I don't understand. Basically my HQ and BR1 phone registered to CUCM and resides in device pool DP-HQ and DP-BR1, I've configured region between HQ and BR1 for G729 only. My MOH server reside in DP-MOH, which have region codec G711 to both HQ and SB. If I don't enable MOH with G729 codec, HQ Phone been put on hold, MOH OK, not before BR1 Phone. I hear Tone On Hold instead. But if I enable MOH with G729, both HQ BR1 phones been put on hold and MOH working fine. In my case, since I already have MOH on different DP and Region codec G711, why would I need to enable G729 for MOH to work on BR1 phones? By the way, I was testing unicast MOH. Regards, Alex ___ For more information regarding industry leading 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 between HQ and BR1 Phones
Guys, Please disregards this message, I found out what's wrong already. Wrong Region configured on MOH DP :P Regards, Alex On Wed, Jun 22, 2011 at 10:21 PM, Alex Goh ncsalex@gmail.com wrote: Hi Guys, I was trying some MOH question just now and notice something which I don't understand. Basically my HQ and BR1 phone registered to CUCM and resides in device pool DP-HQ and DP-BR1, I've configured region between HQ and BR1 for G729 only. My MOH server reside in DP-MOH, which have region codec G711 to both HQ and SB. If I don't enable MOH with G729 codec, HQ Phone been put on hold, MOH OK, not before BR1 Phone. I hear Tone On Hold instead. But if I enable MOH with G729, both HQ BR1 phones been put on hold and MOH working fine. In my case, since I already have MOH on different DP and Region codec G711, why would I need to enable G729 for MOH to work on BR1 phones? By the way, I was testing unicast MOH. Regards, Alex ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] CUE Voiceview Issue
Hi Guys, Anyone encounter this issue before, after voiceview was configured on the CUE and service subscribed to the IP Phone. I've able to login and see number of message. but when I tried to play the message or send a message, I always get Authentication error. Report this error to your system administrator. My CUE is integrate with CUCM and verified correct license file was installed. It seems like something to do with the Authentication URL in the service, anyone can shed some light on this? trying to do some trace on CUE and this is what I've got 4519 06/17 17:00:15.815 vovw cont 0 Enter Controller Requested URI: /voiceview/common/login.do 4519 06/17 17:00:15.815 vovw sydb 0 /sw/apps/vui/vvconfig/enabled 4519 06/17 17:00:15.816 vovw sydb 0 1 4519 06/17 17:00:15.816 vovw cont 0 Host : 142.1.66.253 4519 06/17 17:00:15.817 vovw cont 0 Connection : close 4519 06/17 17:00:15.817 vovw cont 0 User-Agent : Allegro-Software-WebClient/4.34 4519 06/17 17:00:15.817 vovw cont 0 Accept : x-CiscoIPPhone/*, text/*,image/png,*/* 4519 06/17 17:00:15.817 vovw cont 0 Accept-Language : en_US 4519 06/17 17:00:15.817 vovw cont 0 Accept-Charset : utf-8,iso-8859-1;q=0.8 4519 06/17 17:00:15.818 vovw cont 0 x-CiscoIPPhoneModelName : CP-7961G 4519 06/17 17:00:15.818 vovw cont 0 x-CiscoIPPhoneSDKVersion : 7.0.1 4519 06/17 17:00:15.818 vovw cont 0 x-CiscoIPPhoneDisplay : 298,144,3,G 4519 06/17 17:00:15.818 vovw sydb 0 /sw/apps/platformCapabilities/system/preferred_language 4519 06/17 17:00:15.819 vovw sydb 0 en_US 4519 06/17 17:00:15.819 vovw cont 0 Setting session locale en_US 4519 06/17 17:00:15.819 vovw sydb 0 /sw/apps/monitor/ctrl/offline 4519 06/17 17:00:15.820 vovw sydb 0 0 4519 06/17 17:00:15.820 vovw cont 0 Center Controller Requested URI: /voiceview/common/login.do 4519 06/17 17:00:15.821 vovw sess 0 request 4519 06/17 17:00:15.821 vovw sess 0 Querying the phone for its device information. 4519 06/17 17:00:15.953 vovw sess 0 Phone Model : CP-7961G 4519 06/17 17:00:15.953 vovw sess 0 Phone MAC Address: 001E138C3CFC 4519 06/17 17:00:15.953 vovw sess 0 Phone Primary DN : 4001 4519 06/17 17:00:15.953 vovw sess 0 Checking if PIN less login is configured for 4001 4519 06/17 17:00:15.985 VMSS vmdb 0 Request connection: inUse: 0, active: 2 4519 06/17 17:00:15.985 VMSS vmdb 0 Got connection: 0, inUse: 1, active: 2 4519 06/17 17:00:15.985 VMSS vmdb 7 select mailboxid from vm_mbxusers where owner=true and userdn='/sw/local/users/scph1'; 4519 06/17 17:00:15.989 VMSS vmdb 3 PERSONAL_000 4519 06/17 17:00:15.989 VMSS vmdb 0 Freed connection: 0, inUse: 0, active: 2 4519 06/17 17:00:15.990 vovw sess 0 Found mailbox 4519 06/17 17:00:15.990 vovw sess 0 PIN-less login: 0 4519 06/17 17:00:15.990 vovw sess 0 checkPinLess false 4519 06/17 17:00:15.994 vovw cont 0 Exit Controller Requested URI: /voiceview/WEB-INF/screens/phoneobjects/CiscoIPPhoneInput.jsp 4513 06/17 17:00:24.520 vovw cont 0 Enter Controller Requested URI: /voiceview/common/login.do 4513 06/17 17:00:24.520 vovw sydb 0 /sw/apps/vui/vvconfig/enabled 4513 06/17 17:00:24.521 vovw sydb 0 1 4513 06/17 17:00:24.522 vovw cont 0 Submit Type 'LOGIN' 4513 06/17 17:00:24.522 vovw sydb 0 /sw/apps/monitor/ctrl/offline 4513 06/17 17:00:24.523 vovw sydb 0 0 4513 06/17 17:00:24.523 vovw cont 0 Center Controller Requested URI: /voiceview/common/login.do 4513 06/17 17:00:24.524 vovw sess 0 LOGIN request 4513 06/17 17:00:24.539 VMSS vmdb 0 Request connection: inUse: 0, active: 2 4513 06/17 17:00:24.539 VMSS vmdb 0 Got connection: 1, inUse: 1, active: 2 4513 06/17 17:00:24.539 VMSS vmdb 7 select mailboxid from vm_mbxusers where owner=true and userdn='/sw/local/users/scph1'; 4513 06/17 17:00:24.544 VMSS vmdb 3 PERSONAL_000 4513 06/17 17:00:24.544 VMSS vmdb 0 Freed connection: 1, inUse: 0, active: 2 4513 06/17 17:00:24.545 vovw sess 0 4001 4513 06/17 17:00:24.545 vovw sess 0 Found mailbox 4513 06/17 17:00:24.545 vovw sess 0 Valid extension 4513 06/17 17:00:24.545 vovw sess 0 Authenticating user 4513 06/17 17:00:24.545 vovw sess 0 SessionProperties doLogoutCleanup for 4001 PERSONAL_000 4513 06/17 17:00:24.552 vovw sess 0 Personal mailbox locked. Logging him out first 4513 06/17 17:00:24.552 VMSS vmbx 0x01309cd1135a 9 /sw/local/users/scph1 4513 06/17 17:00:24.553 VMSS vmdb 0 Request connection: inUse: 0, active: 2 4513 06/17 17:00:24.553 VMSS vmdb 0 Got connection: 0, inUse: 1, active: 2 4513 06/17 17:00:24.553 VMSS vmdb 7 select vm_message.messageid, vm_message.uid, recent from vm_message, vm_usermsg where state=3 and mailboxid='PERSONAL_000' and vm_message.messageid=vm_usermsg.messageid; 4513 06/17 17:00:24.561 VMSS vmdb 0 Freed connection: 0, inUse: 0, active: 2 4513 06/17 17:00:24.563 vovw sess 0 SessionProperties logged out 4001 session: 34o4vxpr71 4513 06/17 17:00:24.563 vovw sess 0 SessionProperties number of users now: 0 4513 06/17 17:00:24.563 vovw sess 0 Checking if PIN less login is configured for
Re: [OSL | CCIE_Voice] CUE Voiceview Issue
Hi Vinay George, Thanks for pointing that out, I will give it a try. forgot to search thru the archives before posting : ) Alex On Fri, Jun 17, 2011 at 6:51 PM, George Goglidze gogli...@gmail.com wrote: Hi Alex, This has been discussed in numerous ocasions, here's one link to archives: http://onlinestudylist.com/archives/ccie_voice/2010-April/015608.html Regards, On Fri, Jun 17, 2011 at 10:07 AM, Alex Goh ncsalex@gmail.com wrote: Hi Guys, Anyone encounter this issue before, after voiceview was configured on the CUE and service subscribed to the IP Phone. I've able to login and see number of message. but when I tried to play the message or send a message, I always get Authentication error. Report this error to your system administrator. My CUE is integrate with CUCM and verified correct license file was installed. It seems like something to do with the Authentication URL in the service, anyone can shed some light on this? trying to do some trace on CUE and this is what I've got 4519 06/17 17:00:15.815 vovw cont 0 Enter Controller Requested URI: /voiceview/common/login.do 4519 06/17 17:00:15.815 vovw sydb 0 /sw/apps/vui/vvconfig/enabled 4519 06/17 17:00:15.816 vovw sydb 0 1 4519 06/17 17:00:15.816 vovw cont 0 Host : 142.1.66.253 4519 06/17 17:00:15.817 vovw cont 0 Connection : close 4519 06/17 17:00:15.817 vovw cont 0 User-Agent : Allegro-Software-WebClient/4.34 4519 06/17 17:00:15.817 vovw cont 0 Accept : x-CiscoIPPhone/*, text/*,image/png,*/* 4519 06/17 17:00:15.817 vovw cont 0 Accept-Language : en_US 4519 06/17 17:00:15.817 vovw cont 0 Accept-Charset : utf-8,iso-8859-1;q=0.8 4519 06/17 17:00:15.818 vovw cont 0 x-CiscoIPPhoneModelName : CP-7961G 4519 06/17 17:00:15.818 vovw cont 0 x-CiscoIPPhoneSDKVersion : 7.0.1 4519 06/17 17:00:15.818 vovw cont 0 x-CiscoIPPhoneDisplay : 298,144,3,G 4519 06/17 17:00:15.818 vovw sydb 0 /sw/apps/platformCapabilities/system/preferred_language 4519 06/17 17:00:15.819 vovw sydb 0 en_US 4519 06/17 17:00:15.819 vovw cont 0 Setting session locale en_US 4519 06/17 17:00:15.819 vovw sydb 0 /sw/apps/monitor/ctrl/offline 4519 06/17 17:00:15.820 vovw sydb 0 0 4519 06/17 17:00:15.820 vovw cont 0 Center Controller Requested URI: /voiceview/common/login.do 4519 06/17 17:00:15.821 vovw sess 0 request 4519 06/17 17:00:15.821 vovw sess 0 Querying the phone for its device information. 4519 06/17 17:00:15.953 vovw sess 0 Phone Model : CP-7961G 4519 06/17 17:00:15.953 vovw sess 0 Phone MAC Address: 001E138C3CFC 4519 06/17 17:00:15.953 vovw sess 0 Phone Primary DN : 4001 4519 06/17 17:00:15.953 vovw sess 0 Checking if PIN less login is configured for 4001 4519 06/17 17:00:15.985 VMSS vmdb 0 Request connection: inUse: 0, active: 2 4519 06/17 17:00:15.985 VMSS vmdb 0 Got connection: 0, inUse: 1, active: 2 4519 06/17 17:00:15.985 VMSS vmdb 7 select mailboxid from vm_mbxusers where owner=true and userdn='/sw/local/users/scph1'; 4519 06/17 17:00:15.989 VMSS vmdb 3 PERSONAL_000 4519 06/17 17:00:15.989 VMSS vmdb 0 Freed connection: 0, inUse: 0, active: 2 4519 06/17 17:00:15.990 vovw sess 0 Found mailbox 4519 06/17 17:00:15.990 vovw sess 0 PIN-less login: 0 4519 06/17 17:00:15.990 vovw sess 0 checkPinLess false 4519 06/17 17:00:15.994 vovw cont 0 Exit Controller Requested URI: /voiceview/WEB-INF/screens/phoneobjects/CiscoIPPhoneInput.jsp 4513 06/17 17:00:24.520 vovw cont 0 Enter Controller Requested URI: /voiceview/common/login.do 4513 06/17 17:00:24.520 vovw sydb 0 /sw/apps/vui/vvconfig/enabled 4513 06/17 17:00:24.521 vovw sydb 0 1 4513 06/17 17:00:24.522 vovw cont 0 Submit Type 'LOGIN' 4513 06/17 17:00:24.522 vovw sydb 0 /sw/apps/monitor/ctrl/offline 4513 06/17 17:00:24.523 vovw sydb 0 0 4513 06/17 17:00:24.523 vovw cont 0 Center Controller Requested URI: /voiceview/common/login.do 4513 06/17 17:00:24.524 vovw sess 0 LOGIN request 4513 06/17 17:00:24.539 VMSS vmdb 0 Request connection: inUse: 0, active: 2 4513 06/17 17:00:24.539 VMSS vmdb 0 Got connection: 1, inUse: 1, active: 2 4513 06/17 17:00:24.539 VMSS vmdb 7 select mailboxid from vm_mbxusers where owner=true and userdn='/sw/local/users/scph1'; 4513 06/17 17:00:24.544 VMSS vmdb 3 PERSONAL_000 4513 06/17 17:00:24.544 VMSS vmdb 0 Freed connection: 1, inUse: 0, active: 2 4513 06/17 17:00:24.545 vovw sess 0 4001 4513 06/17 17:00:24.545 vovw sess 0 Found mailbox 4513 06/17 17:00:24.545 vovw sess 0 Valid extension 4513 06/17 17:00:24.545 vovw sess 0 Authenticating user 4513 06/17 17:00:24.545 vovw sess 0 SessionProperties doLogoutCleanup for 4001 PERSONAL_000 4513 06/17 17:00:24.552 vovw sess 0 Personal mailbox locked. Logging him out first 4513 06/17 17:00:24.552 VMSS vmbx 0x01309cd1135a 9 /sw/local/users/scph1 4513 06/17 17:00:24.553 VMSS vmdb 0 Request connection: inUse: 0, active: 2 4513 06/17 17:00:24.553 VMSS vmdb 0 Got connection: 0, inUse: 1, active: 2 4513 06/17 17
[OSL | CCIE_Voice] IPPM Add by Extension
Hi Guys, Anyone encounter this issue before? when I try to adding contact in IPPM using the AddByExt options, and it says no UserID matches the extension I've the extension number configured under the End User page, Telephone Number field, also selected the primary extension for the user. I did have the DN's associate with the end user too. I can added the contact using UserID, but not AddByExt, other than this, the IPPM is working fine. Thanks Regards, Alex ___ For more information regarding industry leading 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] DEBUG MGCP PACKET
Hi Randall, See if this help. is about debug of MGCP packet during call preservation for CUCM failover. Alex On Sun, Jun 5, 2011 at 11:01 AM, Randall Crumm rrcr...@yahoo.com wrote: Hi, Does someone have a good example of a debug mgcp packets and brief explanation? RSIP/AUEP/AUCX 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] DEBUG MGCP PACKET
Ops forgot about the url http://ccie-musketeers.blogspot.com/2011/04/mgcp-call-preservation-porcess-with.html On Sun, Jun 5, 2011 at 12:53 PM, Alex Goh ncsalex@gmail.com wrote: Hi Randall, See if this help. is about debug of MGCP packet during call preservation for CUCM failover. Alex On Sun, Jun 5, 2011 at 11:01 AM, Randall Crumm rrcr...@yahoo.com wrote: Hi, Does someone have a good example of a debug mgcp packets and brief explanation? RSIP/AUEP/AUCX 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
[OSL | CCIE_Voice] Ignore Presentation Indicators?
Hi Guys, Anyone know what is the options Ignore Presentation Indicators (internal calls only) does under RDP? reading on the help it sound something to do with call display restriction, but whether it check or unchecked I can't see any different. Regards, Alex ___ For more information regarding industry leading 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] AIM-CUE CF card problem
Hi Rab, Thanks for the info, I wish it is good omen too : D Hi Sam, Thanks, but I guess it might not work, and the card reader can't even detected the card, i guess the CF was toasted :( Anyway look like I've no choice but get it off from ebay, as I don't have extra time to search the Sandisk CF which possible compatible. Regards, Alex On Wed, Jun 1, 2011 at 8:43 AM, Sam Park upperlevelpark...@gmail.comwrote: Alex; You need to re-image your CF with another known good CF. I just did this several weeks ago for a UC500 system. I got another CF from a good system, then I used my linux server to do a bit by bit copy of the CF using 2 USB multi-card readers. If you are not familiar with linux you can use the Ultimate Boot CD (partitioning as well as other utilities). So the hard thing might be getting a known good CF. Sam. On Tue, May 31, 2011 at 5:43 PM, ccieid1ot ccieid...@gmail.com wrote: The mem is your ram, CF is your hd. Try the sandisk CF, I'm sure cisco just oem it from either sandisk or another manufacturer. duy ccie #27737 voice tmobile g2 On May 31, 2011 1:22 PM, Alex Goh ncsalex@gmail.com wrote: Hi Guys, Hope I can seek a little help here, my AIM-CUE 1GB CF card failed on me just 1 week before my exam! I've getting the error of Not a cisco supported CF. Please use cisco supported CF and reinstall the software. System Halted. Anyone know how to solved this issue? I've try to reinstall CUE using the boothelper, but no luck. Possibly the CF card is gone case. A search on google mentioned Cisco AIM-CUE check on the CF Card sector size, else refuse to work. But the used 1GB CF card was asking half the price of the AIM-CUE module /w 1GB CF itself on ebay :( It is anyway I can used on 3rd party CF card? saw it also certain SANDISK CF might work, but I'm not sure it is still able to find in the market now. Also, I notice the router Memory CF (MEM-CF-1GB) is selling cheaper than AIM-CUE-1GBCF, I wonder will it able to use? Any help will be appreciated. Regards, Alex ___ For more information regarding industry leading 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] IPPM on cisco7961 didn't alert
Hi Ki Wi, I've encounter the same issue also, and I solved it by changing the Enterprise Parameters Services URL to IP instead of hostname (Apparently, I miss that part when I reverted my VMware snapshot), remember I saw this solution from OSL discussion before. HTH Cheers, Alex On Tue, May 31, 2011 at 11:25 AM, ShinGei Yong shingei.y...@gmail.comwrote: Frens, If i can recall correctly, that was due to that i missed associate the phone with application user Phone Messenger. You need the phone messenger application user to control the IPPM user. Without the association, the messaging will still work but funny stuff come out, if not wrong Shingei. On Tue, May 31, 2011 at 4:52 AM, Ki Wi kiwi.vo...@gmail.com wrote: Hey, Do you still remember how did you resolve this alert issue? I'm still trying to train myself up in CUPS. Last night, my alert was working, my IPPM login wasn't. Today my IPPM is working but no alert. =( All other components are working. On Sun, Dec 26, 2010 at 12:59 AM, ShinGei Yong shingei.y...@gmail.comwrote: Guys, Pls ignore this mail, has managed to figured out the caused. thanks Shingei. On Sat, Dec 25, 2010 at 4:36 PM, ShinGei Yong shingei.y...@gmail.comwrote: Hi, I've configure the IPPM on cisco 7961 phone, everything works smooth other that the message receive alert. It doesn't ring when there is a mgs come in from CIPC or other IPPM.i've set the audible alert to ON but still got no luck. Another IPPM phone encounter the same issue, so don't think is the phone problem. Any idea? Thanks Shingei. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] AIM-CUE CF card problem
Hi Guys, Hope I can seek a little help here, my AIM-CUE 1GB CF card failed on me just 1 week before my exam! I've getting the error of Not a cisco supported CF. Please use cisco supported CF and reinstall the software. System Halted. Anyone know how to solved this issue? I've try to reinstall CUE using the boothelper, but no luck. Possibly the CF card is gone case. A search on google mentioned Cisco AIM-CUE check on the CF Card sector size, else refuse to work. But the used 1GB CF card was asking half the price of the AIM-CUE module /w 1GB CF itself on ebay :( It is anyway I can used on 3rd party CF card? saw it also certain SANDISK CF might work, but I'm not sure it is still able to find in the market now. Also, I notice the router Memory CF (MEM-CF-1GB) is selling cheaper than AIM-CUE-1GBCF, I wonder will it able to use? Any help will be appreciated. Regards, Alex ___ For more information regarding industry leading 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 Manager and CUE Integration
do you mind to post your config on your HQ and CUE? Alex On Tue, May 17, 2011 at 3:19 AM, Stephen Manuel srman...@bellsouth.netwrote: In my home lab, I have the following 2811 router w/NM-CUE module w/7.0.1 software and CCM license. VM Ware Call Manager 7.0.1 software Router has VWIC-2MFT-T1 cards that are connected to my BR1 and BR2 routers both w/VWIC-1MFT-T1 cards, all are showing multi frame established. HQ router is MGCP controlled and contains the CUE Module. I have CFB and Xcoder resources registered in UCM for the HQ router I’ve rechecked the region, mrgl, location settings in UCM and they appear to be correct. Originally I had the CUE module working when the router was a CME router, however I tried to wipe out the router config and start over. I have all the phone and gateway registered with UCM. I then reinstalled the CUE license and software to make it work with Call Manager vs. CME. I have the CUE ports registered in Call Manager The CUE is showing it’s registered with Call Manager. The issue is when I press the messages button on a phone that has a VM box, I get an immediate fast busy. When I call from another phone and the call rolls to VM same result, immediate fast busy. I’m sore of stumped, I’ve suspected that the issue is Codec related, but I’m unsure how to go about determining that. Any basic guidance would be greatly appreciated. Stephen Manuel ___ For more information regarding industry leading 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] Excellent document on LAN QoS
Thanks Shrini for the link, coincidentally I was reading this also, and also I found another blog post by Joe Astorino that is good reading on the LAN QOS, although it used 3560 and more for CCIE RS exam. Cheers, Alex On Wed, May 18, 2011 at 4:54 PM, Shrini linuxbos...@gmail.com wrote: Just gone through Vik's documentation on LAN QoS , I liked the flow chart. Hopefully it is helpful to you too .. so thought of sharing. http://blog.ipexpert.com/2011/05/16/campus-qos-part-1-classification-and-marking-on-the-catalyst-3750/ Thanks Shrini ___ For more information regarding industry leading 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] busy trigger per button
Hi Voiceboy, This is my understanding. huntstop usually is used on shared ephone-dn to limit the incoming call. if you just wanted to have the next incoming call during an active call, busy trigger will do the trick. whereas for max-call-per button, if it is an octo-line, the default is already 8 source: http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/command/reference/cme_m1ht.html#wp1093876 Cheers, Alex 2011/5/8 voice boy voice...@hotmail.com Hi, I need to ask about to use huntstop channel 1 in srst or to use busy trigger per button 1 So that the calls will be forwarded to voicemail if ot have active call Also do i need to use maximum calls per button 4 to be as cucm while it is in srst ?? 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] Vol2 Lab 2 Supplementary services with GK Trunk
Hi Amit, have to you tried to disable the wait for FE H245 TCS? what is the codec that u configured for the MTP? mind to post ur MTP config on the HQ-RTR here? Regards, Alex On Sat, May 7, 2011 at 9:33 PM, amit batra batraji...@yahoo.com wrote: Hello Everyone. I have finished IPexperts Volume 2 lab 2 .. I managed to finish everything apart from VPIM and Supplementary services ...VPIM is a license issue so not worried .. I have configured a GK trunk between CUCM and HQ router (Gatekeeper) and BR2 router. all endpoints are registered to the gatekeepers. No probs till here .. Call from CUCM to CME and vice versa are working. The problem i am facing is , when i make a call from CUCM phone 5001 (SCCP) to a CME phone 3002 (SCCP) audio works fine. i can press hold button on the CUCM phone. When i do that on CME phone i hear beep. but when i press resume on CUCM phone, CME phone keep's giving that beep sound. when i press hold button on my CME and resume , audio start to flow again.. I have configured software MTP on HQ router. Device pool assigned to the GK-Trunk and this software MTP is the same . On GK-Trunk MRGL is assigned .. Media Termination Point Required (ON) Retry Video Call as Audio (ON) Wait for Far End H.245 Terminal Capability Set (ON) Inbound faststart enabled (ON) when i make a call from any device , i can see that my IOS MTP is invoked and participating in the call .. show sccp connections Am i missing anything here ? or do i need to enable anything else..? I hope i am making some sense.. If the question is not clear please let me know. and 1.30 am i cannot write anything more than this.. Thanks in advance .. Regards Amit ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 6.2)
Hi All, I'm practicing for the question 6.2 in Workbook 2, Lab 1. I manage to have the MeetMe conference setup successfuly, HQ Phn, BR1 PSTN all can dial into MeetMe number too. The H/W Conference Bridge was configured in HQ-RTR, and I can verified it was utilized. The problem comes in when I try to add an ad-hoc participant into the MeetMe Conference Bridge on different region. e.g when MeetMe conference Initiator is HQ PH1, and HQ PH2 joined ad-hoc participant BR1 PH1 which is on G729, the BR1 PH1 will get dropped. I know this is due to codec mismatch issue (verified by changing region from HQ to BR1 as G711, it works fine), but I've transcoder added in both HQ BR1 DP MRGL. It looks like the transcoder doesn't get invoked in this case or do transcoder needed to get this working? since I already have H/W Conference configured. Appreciate if anyone can shed more light on this. Thanks Regards, Alex ___ For more information regarding industry leading 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] Meet Me + Ad-hoc Participant (WB2 Lab1 6.2)
Hi Claude, Thanks for the reply, I do have the hardware conference bridge configured on the router, with support of G729 codec of course. That verify by BR1 Phones can join into meet me conference by dialing the number. Issue come in when one of the meetme participant trying bring in another participant on different codec region into the bridge. Thanks Regards, Alex On Sat, Apr 23, 2011 at 6:59 PM, Friderich Claude cfrider...@netcore.luwrote: Hello, I think you are wrong for this question. You must invoke a conference bridge on the router and thanks to the hardware conference bridge it will support g729 Just add the codec g729r8 in the dspfarm profile conference …. Should work. Of course do not forget to put your conference bridge in the MRG and MRGL of your CCM Regards Claude. *Claude Friderich* *PreSales Support* *[image: ccvp_voice_sm]*** *NETCORE PSF S.A.*** 49 rue du Baerendall B.P.65 L-8201 Mamer Téléphone: 31 33 80-407 Fax: 31 33 80 8-407 GSM: 621 303 616 E-mail: cfrider...@netcore.lu *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Alex Goh *Sent:* samedi 23 avril 2011 12:10 *To:* OSL *Subject:* [OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 6.2) Hi All, I'm practicing for the question 6.2 in Workbook 2, Lab 1. I manage to have the MeetMe conference setup successfuly, HQ Phn, BR1 PSTN all can dial into MeetMe number too. The H/W Conference Bridge was configured in HQ-RTR, and I can verified it was utilized. The problem comes in when I try to add an ad-hoc participant into the MeetMe Conference Bridge on different region. e.g when MeetMe conference Initiator is HQ PH1, and HQ PH2 joined ad-hoc participant BR1 PH1 which is on G729, the BR1 PH1 will get dropped. I know this is due to codec mismatch issue (verified by changing region from HQ to BR1 as G711, it works fine), but I've transcoder added in both HQ BR1 DP MRGL. It looks like the transcoder doesn't get invoked in this case or do transcoder needed to get this working? since I already have H/W Conference configured. Appreciate if anyone can shed more light on this. Thanks Regards, Alex -- This email was Anti Virus checked. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] DHCP Issue with FRTS?
Hi All, Was practicing workbook 2 lab 1, on the question regarding QOS between HQ and BR1/BR2, I've enable auto qos voip trust in BR1 router on the PVC interface. Once the router was reboot, I notice that BR1 IP phones wasn't able to get IP address from the DHCP server, which is CUCM Pub in this case. I've tried removed the frame-relay traffic-shaping command on the BR1 FR physical interface, immediately the phones able to grab IP from DHCP server. Can anyone advise on this? My IOS version is 12.4 (24) T Building configuration... Current configuration : 3601 bytes ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname R2 ! boot-start-marker boot-end-marker ! logging message-counter syslog enable secret 5 $1$BF1c$fGTsUdKaCoeiv8BjWdrw2/ ! no aaa new-model network-clock-participate wic 0 ! ! ! dot11 syslog ip source-route ! ! ip cef ! ! no ip domain lookup no ipv6 cef ! multilink bundle-name authenticated ! ! ! ! isdn switch-type primary-ni ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! voice-card 0 dsp services dspfarm ! ! ! ! ! ! archive log config hidekeys ! ! controller E1 0/0/0 pri-group timeslots 1-3,16 service mgcp ! controller E1 0/0/1 channel-group 0 timeslots 1-31 ! ! class-map match-any AutoQoS-VoIP-RTP-Trust match ip dscp ef class-map match-any AutoQoS-VoIP-Control-Trust match ip dscp cs3 match ip dscp af31 ! ! policy-map AutoQoS-Policy-Trust class AutoQoS-VoIP-RTP-Trust priority 47 compress header ip rtp class AutoQoS-VoIP-Control-Trust bandwidth percent 5 class class-default fair-queue ! ! ! ! ! interface Loopback0 ip address 172.2.254.1 255.255.255.255 ! interface FastEthernet0/0 no ip address duplex auto speed auto ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! interface FastEthernet0/1/0 switchport trunk native vlan 10 switchport mode trunk switchport voice vlan 20 ! interface FastEthernet0/1/1 switchport trunk native vlan 10 switchport mode trunk switchport voice vlan 20 ! interface FastEthernet0/1/2 ! interface FastEthernet0/1/3 ! interface Serial0/0/0:15 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn bind-l3 ccm-manager isdn outgoing display-ie no cdp enable ! interface Serial0/0/1:0 no ip address encapsulation frame-relay * frame-relay traffic-shaping* ! interface Serial0/0/1:0.1 point-to-point description == FR To HQ bandwidth 384 ip address 10.10.11.2 255.255.255.252 frame-relay interface-dlci 101 class AutoQoS-FR-Se0/0/1:0-101 auto qos voip trust ! interface Vlan1 no ip address ! interface Vlan10 ip address 172.2.12.1 255.255.255.0 ! interface Vlan20 ip address 172.2.11.1 255.255.255.0 ip helper-address 172.1.10.10 ! ip forward-protocol nd ip route 0.0.0.0 0.0.0.0 10.10.11.1 no ip http server no ip http secure-server ! ! ! ! map-class frame-relay AutoQoS-FR-Se0/0/1:0-101 frame-relay cir 384000 frame-relay bc 3840 frame-relay be 0 frame-relay mincir 384000 frame-relay fragment 480 service-policy output AutoQoS-Policy-Trust ! ! ! ! ! ! control-plane ! rmon event 3 log trap AutoQoS description AutoQoS SNMP traps for Voice Drops owner AutoQoS rmon alarm 3 cbQosCMDropBitRate.418.3168001 30 absolute rising-threshold 1 3 falling-threshold 0 owner AutoQoS ! ! voice-port 0/0/0:15 ! voice-port 0/2/0 ! voice-port 0/2/1 ! ccm-manager switchback immediate ccm-manager redundant-host 172.1.10.10 ccm-manager mgcp ! mgcp mgcp call-agent 172.1.10.20 service-type mgcp version 0.1 mgcp dtmf-relay voip codec all mode out-of-band mgcp fax t38 ecm mgcp bind control source-interface Loopback0 mgcp bind media source-interface Loopback0 ! mgcp profile default ! sccp local Loopback0 sccp ccm 172.1.10.20 identifier 1 version 7.0 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register BR1-XCODER ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 maximum sessions 3 associate application SCCP ! ! ! ! ! line con 0 line aux 0 line vty 0 4 password cisco login length 0 ! scheduler allocate 2 1000 end ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Device mobility issue.
Any kind soul able to help? On Thu, Apr 14, 2011 at 5:40 PM, Alex Goh ncsalex@gmail.com wrote: Hi All, Understand that when an IP Phone was roaming and physical location of home DP and roaming DP is different, the roaming sensitive setting of the roaming DP will apply to the phone. However, when i moved my BR1 phone to HQ, the View Current Device Mobility Settings of the BR1 phone showing the roaming DP is Not Selected, I believe it shouldn't be that way? Also I notice the Date Time Group on the roaming phone doesn't follow the roaming DP, i thought DTG suppose to be roaming sensitive setting and will apply to the roaming phone? Thanks Regards, Alex ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Device mobility issue.
Any kind soul able to help? On Fri, Apr 15, 2011 at 5:17 PM, Alex Goh ncsalex@gmail.com wrote: Any kind soul able to help? On Thu, Apr 14, 2011 at 5:40 PM, Alex Goh ncsalex@gmail.com wrote: Hi All, Understand that when an IP Phone was roaming and physical location of home DP and roaming DP is different, the roaming sensitive setting of the roaming DP will apply to the phone. However, when i moved my BR1 phone to HQ, the View Current Device Mobility Settings of the BR1 phone showing the roaming DP is Not Selected, I believe it shouldn't be that way? Also I notice the Date Time Group on the roaming phone doesn't follow the roaming DP, i thought DTG suppose to be roaming sensitive setting and will apply to the roaming phone? Thanks Regards, Alex ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] DHCP Issue with FRTS?
Hi George, Thanks for pointed out could be fragmentation issue. I notice my map-class on HQ-RTR was missing the fragment statement. I believe this could be the issue, going to try it out tomorrow. For some reason, that day after I tried to use auto qos for FRF.12 and the map class doesn't created automatically, hence I've manual create map-class which mistakenly left out the fragmentation statement. Thanks again George. Regards, Alex On Sun, Apr 17, 2011 at 1:24 AM, George Goglidze gogli...@gmail.com wrote: Hi Alex, Did you enable it on the other side? It's not because of traffic-shaping itself. It's has probably happened because you have frame-relay fragmentation enabled. DHCP should not be big packets, so it should not get fragmented, but probably .cnf file download was failing from the tftp server. If you did enable it on the other side too, can you post the config of the other router too then? Regards, On Fri, Apr 15, 2011 at 2:26 AM, Alex Goh ncsalex@gmail.com wrote: Hi All, Was practicing workbook 2 lab 1, on the question regarding QOS between HQ and BR1/BR2, I've enable auto qos voip trust in BR1 router on the PVC interface. Once the router was reboot, I notice that BR1 IP phones wasn't able to get IP address from the DHCP server, which is CUCM Pub in this case. I've tried removed the frame-relay traffic-shaping command on the BR1 FR physical interface, immediately the phones able to grab IP from DHCP server. Can anyone advise on this? My IOS version is 12.4 (24) T Building configuration... Current configuration : 3601 bytes version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname R2 ! boot-start-marker boot-end-marker ! logging message-counter syslog enable secret 5 $1$BF1c$fGTsUdKaCoeiv8BjWdrw2/ ! no aaa new-model network-clock-participate wic 0 ! ! ! dot11 syslog ip source-route ! ! ip cef ! ! no ip domain lookup no ipv6 cef ! multilink bundle-name authenticated ! ! ! ! isdn switch-type primary-ni ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! voice-card 0 dsp services dspfarm ! ! ! ! ! ! archive log config hidekeys ! ! controller E1 0/0/0 pri-group timeslots 1-3,16 service mgcp ! controller E1 0/0/1 channel-group 0 timeslots 1-31 ! ! class-map match-any AutoQoS-VoIP-RTP-Trust match ip dscp ef class-map match-any AutoQoS-VoIP-Control-Trust match ip dscp cs3 match ip dscp af31 ! ! policy-map AutoQoS-Policy-Trust class AutoQoS-VoIP-RTP-Trust priority 47 compress header ip rtp class AutoQoS-VoIP-Control-Trust bandwidth percent 5 class class-default fair-queue ! ! ! ! ! interface Loopback0 ip address 172.2.254.1 255.255.255.255 ! interface FastEthernet0/0 no ip address duplex auto speed auto ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! interface FastEthernet0/1/0 switchport trunk native vlan 10 switchport mode trunk switchport voice vlan 20 ! interface FastEthernet0/1/1 switchport trunk native vlan 10 switchport mode trunk switchport voice vlan 20 ! interface FastEthernet0/1/2 ! interface FastEthernet0/1/3 ! interface Serial0/0/0:15 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn bind-l3 ccm-manager isdn outgoing display-ie no cdp enable ! interface Serial0/0/1:0 no ip address encapsulation frame-relay * frame-relay traffic-shaping* ! interface Serial0/0/1:0.1 point-to-point description == FR To HQ bandwidth 384 ip address 10.10.11.2 255.255.255.252 frame-relay interface-dlci 101 class AutoQoS-FR-Se0/0/1:0-101 auto qos voip trust ! interface Vlan1 no ip address ! interface Vlan10 ip address 172.2.12.1 255.255.255.0 ! interface Vlan20 ip address 172.2.11.1 255.255.255.0 ip helper-address 172.1.10.10 ! ip forward-protocol nd ip route 0.0.0.0 0.0.0.0 10.10.11.1 no ip http server no ip http secure-server ! ! ! ! map-class frame-relay AutoQoS-FR-Se0/0/1:0-101 frame-relay cir 384000 frame-relay bc 3840 frame-relay be 0 frame-relay mincir 384000 frame-relay fragment 480 service-policy output AutoQoS-Policy-Trust ! ! ! ! ! ! control-plane ! rmon event 3 log trap AutoQoS description AutoQoS SNMP traps for Voice Drops owner AutoQoS rmon alarm 3 cbQosCMDropBitRate.418.3168001 30 absolute rising-threshold 1 3 falling-threshold 0 owner AutoQoS ! ! voice-port 0/0/0:15 ! voice-port 0/2/0 ! voice-port 0/2/1 ! ccm-manager switchback immediate ccm-manager redundant-host 172.1.10.10 ccm-manager mgcp ! mgcp mgcp call-agent 172.1.10.20 service-type mgcp version 0.1 mgcp dtmf-relay voip codec all mode out-of-band mgcp fax t38 ecm mgcp bind control source-interface Loopback0 mgcp bind media source-interface Loopback0 ! mgcp profile default ! sccp local Loopback0
Re: [OSL | CCIE_Voice] Device mobility issue.
Hi Vik, The device mobility is on for the phone, and the subnet is correctly attached for BR1 and HQ site. I'm will try to redo the DMI see how it goes. Thanks Regards, Alex On Sun, Apr 17, 2011 at 11:38 AM, Vik Malhi vma...@ipexpert.com wrote: Either roaming is not enabled for the phone. Or you have not attached the subnet of the HQ site to the HQ device pool (using device mobility info under the system menu). -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.comvma...@ipexpert.com Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Live Assistance, Please visit: http://www.ipexpert.com/chat www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Wireless, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ On Apr 16, 2011, at 20:22, Alex Goh ncsalex@gmail.com wrote: Any kind soul able to help? On Fri, Apr 15, 2011 at 5:17 PM, Alex Goh ncsalex@gmail.com ncsalex@gmail.com wrote: Any kind soul able to help? On Thu, Apr 14, 2011 at 5:40 PM, Alex Goh ncsalex@gmail.com ncsalex@gmail.com wrote: Hi All, Understand that when an IP Phone was roaming and physical location of home DP and roaming DP is different, the roaming sensitive setting of the roaming DP will apply to the phone. However, when i moved my BR1 phone to HQ, the View Current Device Mobility Settings of the BR1 phone showing the roaming DP is Not Selected, I believe it shouldn't be that way? Also I notice the Date Time Group on the roaming phone doesn't follow the roaming DP, i thought DTG suppose to be roaming sensitive setting and will apply to the roaming phone? Thanks Regards, Alex ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.comwww.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Device mobility issue.
Hi All, Understand that when an IP Phone was roaming and physical location of home DP and roaming DP is different, the roaming sensitive setting of the roaming DP will apply to the phone. However, when i moved my BR1 phone to HQ, the View Current Device Mobility Settings of the BR1 phone showing the roaming DP is Not Selected, I believe it shouldn't be that way? Also I notice the Date Time Group on the roaming phone doesn't follow the roaming DP, i thought DTG suppose to be roaming sensitive setting and will apply to the roaming phone? Thanks Regards, Alex ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 4.2)
Hi All, Thanks very much for the reply. The issue is due to my mistake that registering BR2 to wrong zone. Now the CUCM Call to BR2 is working fine except the supplementary service e.g hold, Moh doesn't work, do I need MTP for this? also, calling from BR2 Sip phone to CUCM is failling, phone ring, but when answered, it dropped. my Sip phone is using G729 codec, do I still need MTP on BR2 in this case? Thanks Regards, Alex On Sun, Apr 10, 2011 at 2:19 AM, Naoufal Kerboute naou...@mhdinfotech.com wrote: Hi, You have to register the br2 with the UCME zone not the VIA zone. Remove h323-gateway voip id VIA ipaddr 172.1.254.1 1719 and replace it with h323-gateway voip id UCME ipaddr 172.1.254.1 1719 Thanks Naoufal -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Alex Goh Sent: Saturday, April 09, 2011 9:43 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 4.2) Hi Guys, I'm trying to get the solutions for question 4.2 to work, but apparently I'm missing something and hope someone can help. I've search thru the list but doesn't really found a solution work for my case. The issue I've encounter are when HQ phone 5001 calling BR2 phone 3003, 3003 ring, but when i tried to answered, the call drop. I know it might be related to codec issue, but I've my HQ-RTR configured with Xcoder which it is up and active but the call still failing. I also did have the trunk in cucm Wait for Far End H.245 Terminal Capability Set unchecked. once things I notice is that, my call doesn't seems get re-originated on the cube router to BR2 router, what I see during ringing state my show gatekeeper endpoint show the call is directly from the CUCM to BR2 It is only 2 call legs instead of 4 (see below). hm, what have I missed? Some Info: HQ Router (R1) interface Loopback0 ip address 172.1.254.1 255.255.255.255 h323-gateway voip interface h323-gateway voip id VIA ipaddr 172.1.254.1 1719 h323-gateway voip h323-id R1 h323-gateway voip bind srcaddr 172.1.254.1 gatekeeper zone local UCM 172.1.254.1 zone local UCME outvia VIA zone local VIA zone prefix UCME 3... gw-type-prefix 1#* default-technology no shutdown dial-peer voice 30 voip destination-pattern 3... session target ras codec g711ulaw ! dial-peer voice 31 voip incoming called-number 3... Total number of active calls = 1. GATEKEEPER CALL INFO LocalCallID Age(secs) BW 511-32797 6 16(Kbps) Endpt(s): Alias E.164Addr src EP: gk_trunk_2 5001 CallSignalAddr Port RASSignalAddr Port 172.1.10.20 38233 172.1.10.20 32795 Endpt(s): Alias E.164Addr dst EP: R3 3003 CallSignalAddr Port RASSignalAddr Port 172.3.254.1 1720 172.3.254.1 49395 GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags --- - --- - - - 172.1.10.10 47142 172.1.10.10 32838 UCM VOIP-GW H323-ID: gk_trunk_1 Voice Capacity Max.= Avail.= Current.= 0 172.1.10.20 38233 172.1.10.20 32795 UCM VOIP-GW H323-ID: gk_trunk_2 Voice Capacity Max.= Avail.= Current.= 0 172.1.254.1 1720 172.1.254.2 56974 VIA H323-GW H323-ID: R1 Voice Capacity Max.= Avail.= Current.= 0 172.3.254.1 1720 172.3.254.1 49395 VIA H323-GW H323-ID: R3 Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 4 R1(config-if)#do sh gatek gw GATEWAY TYPE PREFIX TABLE = Prefix: 1#* (Default gateway-technology) Zone UCM master gateway list: 172.1.10.20:38233 gk_trunk_2 172.1.10.10:47142 gk_trunk_1 Zone VIA master gateway list: 172.3.254.1:1720 R3 172.1.254.2:1720 R1 BR2 Router (R2) interface Loopback0 ip address 172.3.254.1 255.255.255.255 h323-gateway voip interface h323-gateway voip id VIA ipaddr 172.1.254.1 1719 h323-gateway voip h323-id R3 h323-gateway voip tech-prefix 1# h323-gateway voip bind srcaddr 172.3.254.1 dial-peer voice 10 voip incoming called-number 3... dtmf-relay rtp-nte codec g711ulaw ! CUCM Trunk the trunk was assign a separate DP with a region that using G729 when calling HQ and BR2. Regards, Alex ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 4.2)
Hi Guys, I'm trying to get the solutions for question 4.2 to work, but apparently I'm missing something and hope someone can help. I've search thru the list but doesn't really found a solution work for my case. The issue I've encounter are when HQ phone 5001 calling BR2 phone 3003, 3003 ring, but when i tried to answered, the call drop. I know it might be related to codec issue, but I've my HQ-RTR configured with Xcoder which it is up and active but the call still failing. I also did have the trunk in cucm Wait for Far End H.245 Terminal Capability Set unchecked. once things I notice is that, my call doesn't seems get re-originated on the cube router to BR2 router, what I see during ringing state my show gatekeeper endpoint show the call is directly from the CUCM to BR2 It is only 2 call legs instead of 4 (see below). hm, what have I missed? Some Info: HQ Router (R1) interface Loopback0 ip address 172.1.254.1 255.255.255.255 h323-gateway voip interface h323-gateway voip id VIA ipaddr 172.1.254.1 1719 h323-gateway voip h323-id R1 h323-gateway voip bind srcaddr 172.1.254.1 gatekeeper zone local UCM 172.1.254.1 zone local UCME outvia VIA zone local VIA zone prefix UCME 3... gw-type-prefix 1#* default-technology no shutdown dial-peer voice 30 voip destination-pattern 3... session target ras codec g711ulaw ! dial-peer voice 31 voip incoming called-number 3... Total number of active calls = 1. GATEKEEPER CALL INFO LocalCallIDAge(secs) BW 511-32797 6 16(Kbps) Endpt(s): Alias E.164Addr src EP: gk_trunk_25001 CallSignalAddr Port RASSignalAddr Port 172.1.10.20 38233 172.1.10.20 32795 Endpt(s): Alias E.164Addr dst EP: R33003 CallSignalAddr Port RASSignalAddr Port 172.3.254.1 1720 172.3.254.1 49395 GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags --- - --- - - - 172.1.10.10 47142 172.1.10.10 32838 UCM VOIP-GW H323-ID: gk_trunk_1 Voice Capacity Max.= Avail.= Current.= 0 172.1.10.20 38233 172.1.10.20 32795 UCM VOIP-GW H323-ID: gk_trunk_2 Voice Capacity Max.= Avail.= Current.= 0 172.1.254.1 1720 172.1.254.2 56974 VIA H323-GW H323-ID: R1 Voice Capacity Max.= Avail.= Current.= 0 172.3.254.1 1720 172.3.254.1 49395 VIA H323-GW H323-ID: R3 Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 4 R1(config-if)#do sh gatek gw GATEWAY TYPE PREFIX TABLE = Prefix: 1#*(Default gateway-technology) Zone UCM master gateway list: 172.1.10.20:38233 gk_trunk_2 172.1.10.10:47142 gk_trunk_1 Zone VIA master gateway list: 172.3.254.1:1720 R3 172.1.254.2:1720 R1 BR2 Router (R2) interface Loopback0 ip address 172.3.254.1 255.255.255.255 h323-gateway voip interface h323-gateway voip id VIA ipaddr 172.1.254.1 1719 h323-gateway voip h323-id R3 h323-gateway voip tech-prefix 1# h323-gateway voip bind srcaddr 172.3.254.1 dial-peer voice 10 voip incoming called-number 3... dtmf-relay rtp-nte codec g711ulaw ! CUCM Trunk the trunk was assign a separate DP with a region that using G729 when calling HQ and BR2. Regards, Alex ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 4.2)
Hi ShinGei, Thanks for much for the reply, I guess I must be overlooked, the BR2 was registered to the wrong zone! I'm going to try it on my lab tomorrow. Again thanks! Regards, Alex On Sun, Apr 10, 2011 at 2:18 AM, ShinGei Yong shingei.y...@gmail.com wrote: Hi Alex@ncs, While observing your config,i noticed that you've 3 zone defined under GK, which are UCM,UCME VIA. If i remember correctly,ur R3 which is ur CME site should registered to UCME instead of zone VIA right? Also, what is your region configuration on that pointed to GK? Thanks Shingei. On Sun, Apr 10, 2011 at 1:42 AM, Alex Goh ncsalex@gmail.com wrote: Hi Guys, I'm trying to get the solutions for question 4.2 to work, but apparently I'm missing something and hope someone can help. I've search thru the list but doesn't really found a solution work for my case. The issue I've encounter are when HQ phone 5001 calling BR2 phone 3003, 3003 ring, but when i tried to answered, the call drop. I know it might be related to codec issue, but I've my HQ-RTR configured with Xcoder which it is up and active but the call still failing. I also did have the trunk in cucm Wait for Far End H.245 Terminal Capability Set unchecked. once things I notice is that, my call doesn't seems get re-originated on the cube router to BR2 router, what I see during ringing state my show gatekeeper endpoint show the call is directly from the CUCM to BR2 It is only 2 call legs instead of 4 (see below). hm, what have I missed? Some Info: HQ Router (R1) interface Loopback0 ip address 172.1.254.1 255.255.255.255 h323-gateway voip interface h323-gateway voip id VIA ipaddr 172.1.254.1 1719 h323-gateway voip h323-id R1 h323-gateway voip bind srcaddr 172.1.254.1 gatekeeper zone local UCM 172.1.254.1 zone local UCME outvia VIA zone local VIA zone prefix UCME 3... gw-type-prefix 1#* default-technology no shutdown dial-peer voice 30 voip destination-pattern 3... session target ras codec g711ulaw ! dial-peer voice 31 voip incoming called-number 3... Total number of active calls = 1. GATEKEEPER CALL INFO LocalCallID Age(secs) BW 511-32797 6 16(Kbps) Endpt(s): Alias E.164Addr src EP: gk_trunk_2 5001 CallSignalAddr Port RASSignalAddr Port 172.1.10.20 38233 172.1.10.20 32795 Endpt(s): Alias E.164Addr dst EP: R3 3003 CallSignalAddr Port RASSignalAddr Port 172.3.254.1 1720 172.3.254.1 49395 GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags --- - --- - - - 172.1.10.10 47142 172.1.10.10 32838 UCM VOIP-GW H323-ID: gk_trunk_1 Voice Capacity Max.= Avail.= Current.= 0 172.1.10.20 38233 172.1.10.20 32795 UCM VOIP-GW H323-ID: gk_trunk_2 Voice Capacity Max.= Avail.= Current.= 0 172.1.254.1 1720 172.1.254.2 56974 VIA H323-GW H323-ID: R1 Voice Capacity Max.= Avail.= Current.= 0 172.3.254.1 1720 172.3.254.1 49395 VIA H323-GW H323-ID: R3 Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 4 R1(config-if)#do sh gatek gw GATEWAY TYPE PREFIX TABLE = Prefix: 1#* (Default gateway-technology) Zone UCM master gateway list: 172.1.10.20:38233 gk_trunk_2 172.1.10.10:47142 gk_trunk_1 Zone VIA master gateway list: 172.3.254.1:1720 R3 172.1.254.2:1720 R1 BR2 Router (R2) interface Loopback0 ip address 172.3.254.1 255.255.255.255 h323-gateway voip interface h323-gateway voip id VIA ipaddr 172.1.254.1 1719 h323-gateway voip h323-id R3 h323-gateway voip tech-prefix 1# h323-gateway voip bind srcaddr 172.3.254.1 dial-peer voice 10 voip incoming called-number 3... dtmf-relay rtp-nte codec g711ulaw ! CUCM Trunk the trunk was assign a separate DP with a region that using G729 when calling HQ and BR2. Regards, Alex ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Multicast MoH to CUBE
Anyone can enlighten me? On Sun, Mar 27, 2011 at 8:23 PM, Alex Goh ncsalex@gmail.com wrote: Hi All, When trying to practice some MoH lab today, I notice that whenever HQ/BR1 phone make a call to BR2 Phone via H323 gateway, which BR2 setup as CME (telephony-service), when HQ/BR1 phone press hold, the MoH will not work (BR2 phone hear silence). But when BR2 phone press Hold, HQ/BR1 phone can hear MoH. Same for when HQ/BR1 tried to send call out to PSTN via the H323 gateway, the PSTN hear no MoH. BR2 Phone call PSTN the MoH is ok though. My HQ is set to send Unicast MoH, while BR1 is set to send Multicast MoH. MoH servers have different DP and Region to all site is g711. I did tried to turn the BR2 into a H323 gateway only (no CUBE) and configured the following commands: ccm-manager music-on-hold, moh music-on-hold.au multicast moh 239.1.1.5 port 16384 Then MoH is working fine when HQ/BR1 put the call to PSTN on hold via the H323 gateway, also when HQ/BR1 calling BR2 Phone which on different DP/Region, MoH is working fine for both direction. Is this normal that Multicast MoH or Unicast MoH is not supporting CUBE in this case? also, why is BR2 CME phone calling HQ/BR1 phone yet MoH is working fine? Regards, Alex ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Multicast MoH to CUBE
Hi All, When trying to practice some MoH lab today, I notice that whenever HQ/BR1 phone make a call to BR2 Phone via H323 gateway, which BR2 setup as CME (telephony-service), when HQ/BR1 phone press hold, the MoH will not work (BR2 phone hear silence). But when BR2 phone press Hold, HQ/BR1 phone can hear MoH. Same for when HQ/BR1 tried to send call out to PSTN via the H323 gateway, the PSTN hear no MoH. BR2 Phone call PSTN the MoH is ok though. My HQ is set to send Unicast MoH, while BR1 is set to send Multicast MoH. MoH servers have different DP and Region to all site is g711. I did tried to turn the BR2 into a H323 gateway only (no CUBE) and configured the following commands: ccm-manager music-on-hold, moh music-on-hold.au multicast moh 239.1.1.5 port 16384 Then MoH is working fine when HQ/BR1 put the call to PSTN on hold via the H323 gateway, also when HQ/BR1 calling BR2 Phone which on different DP/Region, MoH is working fine for both direction. Is this normal that Multicast MoH or Unicast MoH is not supporting CUBE in this case? also, why is BR2 CME phone calling HQ/BR1 phone yet MoH is working fine? Regards, Alex ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MVA Hairpin no Audio
Hi, Not sure is this help, but you might want to turn on MTP on ur h323 gateway? i remember i read it from cisco notes saying it need to be turn on. Regards, Alex On Fri, Mar 25, 2011 at 1:38 AM, study2b ccie study4ccievo...@gmail.com wrote: Hi experts, I had configured MVA using hairpin method. Everything worked and calls went out, but when I picked it up, there were no audio! Has anyone seen this problem before? Where should I start to troubleshoot? FYI, both of dial-peers voip are using no vad and G711ulaw. I can see calls went out on mgcp trunk. Thank you, ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] H323 gateway doesn't send the plus onoutgoing PRI calls
hi i believe it will be more appropriate to add it on voice port if u sending out to pstn, except you are sending it to sip dial-peer, otherwise if h323 it wil get strip off before sending to session target. correct me if i'm wrong. cheers, Alex On Thu, Mar 24, 2011 at 11:28 PM, cciefo...@hotmail.com wrote: Does this have to go on the voice port or can it go on the dial peer too? -Original Message- From: adam compton com...@gmail.com Sender: ccie_voice-boun...@onlinestudylist.com Date: Thu, 24 Mar 2011 08:24:41 To: Bill Lakewhl...@gmail.com Cc: ccie_voice@onlinestudylist.com; Adam Thompsonphoe...@fatturtle.com Subject: Re: [OSL | CCIE_Voice] H323 gateway doesn't send the plus on outgoing PRI calls ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
