[OSL | CCIE_Voice] Proctor Lab voice rack rental vouchers for sale
Hello List If any of you are interested in buying proctor labs vouchers for your voice practice, I have got a bunch I will not going to use anymore and I'm sure we can work out a fair price on it. Please unicast me if interested Best regards Daniel ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUE Integration Commands
Check the configuration examples and technotes under CUE in the support page. The first one is about CUCM to CUE integration. It starts with the GUI method and in the end of the doco you will find the cli portion you are after. On Sat, Oct 23, 2010 at 9:26 AM, ccieiwillb ccieiwi...@gmail.com wrote: Thanks ShinGei, But I was looking for the document with the manual commands entered on CUE. The ones you enter under cnn system to configure Jtapi user, cti ports and callmanager ip address. On Fri, Oct 22, 2010 at 12:34 PM, ShinGei Yong shingei.y...@gmail.comwrote: Hi, Here you go... http://www.cisco.com/en/US/products/sw/voicesw/ps5520/products_configuration_example09186a0080289ef0.shtml Ensure you have correct license file for CUE. TIA Shingei On Fri, Oct 22, 2010 at 11:09 PM, ccieiwillb ccieiwi...@gmail.comwrote: Hi Experts, I was wondering if anyone happened to have a link to the commands to manually integrate CUE with CUCM? I have seen the document before off of the Cisco website but I am not able to find it any longer. I am having a strange issue whenever I launch the initiaulization webpage so I am trying to see if I can configure the CTI ports and jtapi manually to integrate with CUCM. Any help is greatyly appreciated. Thanks ccieiwillb ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME SIP phone - call-forward all to VM
what does debug ccsip messages show you? If you ring from CME to CUE does it work? Can you provide a bit more info? Cheers On Tue, Oct 19, 2010 at 1:11 PM, David A david.a...@gmail.com wrote: Hi All, I was working on a scenario where I need a number 1003 on a SIP CME call-forward all to voicemail. I created a voice register dn with number 1003 and call-forwaded to 1600 the voicemail pilot. When I dial this number I get fastbusy and debug shows no dial-peer with 1003 and there is no dial-peer with 1003 in show dial-peer voice summ. What am I missing? Thanks, DA ___ 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] Call Forward Unregistered
Hello guys If you want to manipulate this with CUCM the place to change the redirected number is the VM profile as indicated by Mark. Alternatively you could attach an additional rule to the translation-profile plugged inbound to the POTS call leg in the branch router in SRST mode and configure it to change the redirect-called number from to the e164 that you are after. Cheers On Mon, Oct 18, 2010 at 11:36 AM, Mark Holloway m...@markholloway.com wrote: I ran into this same issue. Supposedly it's a problem with CUCM 7.0 and VMWare. If you go to the Device Phone and click on the Site B phones Line and specifically assign the Voicemail Profile to the Line it might work. I had success a couple of times doing this, but then after resetting my rack the last time and assigning the VM profile to the Line I still had this issue. On Oct 17, 2010, at 3:28 PM, Afzal Bhutta wrote: Scenario: In SRST mode: HQ MGCP gateway and Site-B H322 gateway Site-C H322 gateway cme HQ and Site C phones are being able to call SiteB Phone 1 using 4 digits dialing in SRST.(Wan failure) I use call forward unregistered feature. When I call from HQ Phone-1 call routed through HQ Gateway. When I call from Site-C Phone-1 call routed through the GK first and then HQ Gateway. Below is the display I am getting on my Site-B phone display. Forward HQ Phone 1 (2001) For 3001 By3001 Forward Site-C Phone 1 (4001) For 3001 By3001 My question how can I achieve below display in FOR and BY field it should be E.164 number format and than 4 digits internal ID Forward (2001) For +19723033001 (3...) By+19723033001 (3...) Forward (4001) For +19723033001 (3...) By+19723033001 (3...) Thanking you in anticipation folks. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 2 Lab2 IPMA Problem
do you have call forward settings in cti route point to point to mgr dn + css configured? On Sat, Oct 16, 2010 at 1:58 PM, Ryan Schwab schwab...@shaw.ca wrote: Hi Guys, Working on question 10.1 on Lab2. I have IPMA configured with Extension mobility and everything seems to be working fine except for one big problem. When I turn off DivertAll on the managers phone, and I call the managers extension (1080), I expect this to wring directly to the managers phone. However, it doesn’t. Calls from internal phones or PSTN phone attempts to make the call, but nothing happens and after a few seconds releases the call. Like I said, everything else is working. I can intercept calls, but can just not ring the manager phone directly. Anyone run into something like this? ___ 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] Voicemail Speed-dial Button for SIP in SRST mode?
I have to test this but in SIP SRST the phones will send its configuration files to SRST and not the other way around. So you may not need to worry about the voicemail speed dial in SRST if that setting was downloaded to the phone from CUCM in the first place. Please test and let us know. Cheers On Thu, Oct 14, 2010 at 1:14 AM, Tam Nhu tamnhu...@gmail.com wrote: Might be I am wrong, but I don't see there is a way to set up the 'voicemail' speed-dial button for SIP phones in SRST mode with the lab IOS version 12.4(20)T2. There is no such the command 'voicemail' in SRST mode for SIP (as it does for SCCP), just simply 'max-dn' and 'max-pool' under 'voice register global'. Call forwarding is configured under 'voice register pool', but it is really no such command for 'voicemail' button anywhere. After a few hours of searching and trying to find a relevant match for this topic, but I could not find anything related. There is no such requirement in any IPX Vol 1 and 2 labs as well. If someone came across this and know the work-around, please share. Or just please to confirm that there is no way. I just think up anything that they might ask to do in the real lab. Thanks, TN. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME Transcoder for CUE
Remove voice-class codec from you inbound voip dial-peer to force a mismatch. On Wed, Oct 13, 2010 at 2:28 PM, Warren Heaviside (wheavisi) wheav...@cisco.com wrote: I’m having trouble invoking a Transcoding resource for an HQ to BR2-CUE call. When calling HQ to BR2 I’ve verified the call is established using G729. When RNA forwarding to CUE it goes fast busy due to not invoking a Transcoder for G729/G711. I can’t see what’s missing below. I’ve included a show sccp at the bottom. Thanks, Warren dspfarm dsp services dspfarm ! voice-card 2 ! ! ! voice service voip clid network-provided allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip fax protocol cisco h323 h225 timeout setup 3 no h225 timeout keepalive sip bind control source-interface GigabitEthernet0/0 bind media source-interface GigabitEthernet0/0 ! ! ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g711alaw codec preference 3 g729r8 codec preference 4 g729br8 ! ! ! ! voice class h323 1 h225 timeout tcp establish 3 ! ! ! ! voice class custom-cptone leave dualtone conference frequency 700 cadence 300 250 ! voice class custom-cptone join dualtone conference frequency 900 cadence 300 50 300 50 ! ! ! ! ! ! ! voice register global system message SRST mode in effect ! ! voice translation-rule 852 rule 1 /.*\(4...\)/ /\1/ ! voice translation-rule 999 rule 1 // // type any national plan any unknown ! ! voice translation-profile in translate called 852 ! voice translation-profile national translate calling 999 translate called 999 ! ! ! ! ! ! username wheavisi password 0 htts123 archive log config hidekeys ! ! ! ! ! ! ! ! ! interface GigabitEthernet0/0 ip address 172.16.184.63 255.255.255.128 duplex auto speed auto media-type rj45 h323-gateway voip interface h323-gateway voip id GK ipaddr 172.16.184.78 1719 h323-gateway voip h323-id CME h323-gateway voip tech-prefix 852 h323-gateway voip bind srcaddr 172.16.184.63 ! interface GigabitEthernet0/1 no ip address shutdown duplex auto speed auto media-type rj45 ! interface Content-Engine1/0 no ip address shutdown ! interface Service-Engine4/0 ip unnumbered GigabitEthernet0/0 service-module ip address 172.16.184.64 255.255.255.128 service-module ip default-gateway 172.16.184.63 no keepalive ! ip forward-protocol nd ip route 0.0.0.0 0.0.0.0 172.16.184.1 ip route 172.16.184.64 255.255.255.255 Service-Engine4/0 ip http server ip http authentication local no ip http secure-server ip http path flash:gui ! ! ! no logging trap ! ! ! ! ! tftp-server flash:P00307010200.bin tftp-server flash:P00307010200.loads tftp-server flash:P00307010200.sb2 tftp-server flash:P00307010200.sbn tftp-server flash:Desktops/320x212x16/SME_IP7965.png tftp-server flash:Desktops/320x212x16/List.xml tftp-server flash:Desktops/320x212x16/SME_IP7965_Thumbnail.png ! control-plane ! ! ! voice-port 0/0/0 ! voice-port 0/0/1 ! ! mgcp fax t38 ecm ! sccp local GigabitEthernet0/0 sccp ccm 172.16.184.63 identifier 1 version 7.0 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register htts-transcoder associate profile 2 register htts-conf ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 2 associate application SCCP ! dial-peer voice 911 pots translation-profile outgoing 911 destination-pattern 911 clid strip name no digit-strip ! dial-peer voice 999 pots translation-profile outgoing 8digitANI destination-pattern 999 progress_ind setup enable 3 ! dial-peer voice 1 voip voice-class codec 1 voice-class h323 1 incoming called-number . ! dial-peer voice 5009 voip preference 1 destination-pattern 5009 voice-class codec 1 voice-class h323 1 session target ipv4:172.16.184.135 dtmf-relay h245-alphanumeric h245-signal ip qos dscp cs3 signaling ! dial-peer voice 5 voip destination-pattern [23]... voice-class codec 1 voice-class h323 1 session target ras tech-prefix 2# ! dial-peer voice 6 voip translation-profile incoming in voice-class codec 1 voice-class h323 1 session target ras incoming called-number 852. ! dial-peer voice 7 voip destination-pattern 5009 voice-class codec 1 voice-class h323 1 session target ras tech-prefix 2# ! dial-peer voice 2300 voip preference 1 destination-pattern [23]... voice-class codec 1 session target ipv4:172.16.184.135 dtmf-relay h245-alphanumeric h245-signal ip qos dscp cs3
Re: [OSL | CCIE_Voice] MVA Troubleshooting lab 6 question 5.3
do a partial match or deploy your translation-profile to your pots inbound dial-peer dial-peer voice mvanumber pots service mva translation-profile incoming bla bla in you translation-profile you change the ANI from whatever it is not matching the remote destination to what matches Alternatively you can do partial match and you will need to do and re-do the service parameter configuraiton a few times until it takes effect. very buggy in this version of cucm. On Thu, Oct 7, 2010 at 10:52 AM, Pithog Oil pithog...@yahoo.com wrote: I think my challenge is this, how do i apply (the translation rule) to get my MVA number match. *avid Lee d16...@gmail.com* wrote: From: David Lee d16...@gmail.com Subject: MVA Troubleshooting lab 6 question 5.3 To: ccie_voice@onlinestudylist.com Cc: pithog...@yahoo.com Date: Wednesday, October 6, 2010, 9:40 PM Hi there, Make sure that the MVA number under Media resources matches the actual number passed to UCM from the dial-peer. (i.e. check your voice translation rules.) The prompt will play whether that matches or not, but once passed to UCM, the DNIS that UCM sees has to match the MVA DN. Thanks, -Dave On Wed, Oct 6, 2010 at 5:29 PM, ccie_voice-requ...@onlinestudylist.comhttp://us.mc1204.mail.yahoo.com/mc/compose?to=ccie_voice-requ...@onlinestudylist.com wrote: Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.comhttp://us.mc1204.mail.yahoo.com/mc/compose?to=ccie_vo...@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.comhttp://us.mc1204.mail.yahoo.com/mc/compose?to=ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.comhttp://us.mc1204.mail.yahoo.com/mc/compose?to=ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. MVA Troubleshooting lab 6 question 5.3 (Pithog Oil) 2. Re: UCCX challenges (Tamer Ismail) 3. Re: UCCX challenges (cciefo...@hotmail.comhttp://us.mc1204.mail.yahoo.com/mc/compose?to=cciefo...@hotmail.com ) 4. Re: UCCX challenges (Pithog Oil) 5. Re: UCCX challenges (Warren Heaviside (wheavisi)) -- Message: 1 Date: Wed, 6 Oct 2010 14:24:17 -0700 (PDT) From: Pithog Oil pithog...@yahoo.comhttp://us.mc1204.mail.yahoo.com/mc/compose?to=pithog...@yahoo.com To: ccie_voice@onlinestudylist.comhttp://us.mc1204.mail.yahoo.com/mc/compose?to=ccie_vo...@onlinestudylist.com Subject: [OSL | CCIE_Voice] MVA Troubleshooting lab 6 question 5.3 Message-ID: 148106.91353...@web120414.mail.ne1.yahoo.comhttp://us.mc1204.mail.yahoo.com/mc/compose?to=148106.91353...@web120414.mail.ne1.yahoo.com Content-Type: text/plain; charset=iso-8859-1 ? I spent some time trying to figure out a fix but, i have not gotten the solution yet, whenever i call 2123945010 in an attemp to invoke my MVA application, it rings quite fine and prompts me for my remote destination, ID and when i press 1 to call an extension, i try to place a call but the call gets dropped. ? I have my MVA number specified on UCM to be 5010, i will appreciate assistance on how to fix thas issue, ? i think? a translation profile was?used?in ?the solutions to translate /5002/ /2123942123/ but its not clear how?the translation pattern was invoked. ? ?Thanks in Anticipation. ? ? -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20101006/b64eb9de/attachment-0001.html -- Message: 2 Date: Wed, 6 Oct 2010 23:23:43 +0200 From: Tamer Ismail tih...@gmail.comhttp://us.mc1204.mail.yahoo.com/mc/compose?to=tih...@gmail.com To: 'Pithog Oil' pithog...@yahoo.comhttp://us.mc1204.mail.yahoo.com/mc/compose?to=pithog...@yahoo.com Cc: ccie_voice@onlinestudylist.comhttp://us.mc1204.mail.yahoo.com/mc/compose?to=ccie_vo...@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] UCCX challenges Message-ID: 00a801cb659c$c28ac910$47a05b...@com Content-Type: text/plain; charset=us-ascii Hello Pithoq, That's mean script error, it recommend to upload and replace the scripts files on flash. Tamer, From: ccie_voice-boun...@onlinestudylist.comhttp://us.mc1204.mail.yahoo.com/mc/compose?to=ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.comhttp://us.mc1204.mail.yahoo.com/mc/compose?to=ccie_voice-boun...@onlinestudylist.com] On Behalf Of Pithog Oil Sent: Wednesday, October 06, 2010 11:12 PM To: ccie_voice@onlinestudylist.comhttp://us.mc1204.mail.yahoo.com/mc/compose?to=ccie_vo...@onlinestudylist.com Subject: [OSL |
Re: [OSL | CCIE_Voice] MoH SRST (Stream from Flash)`
Check that your voip dial-peers facing the CUCM leg have voice-class codec configured with g729 and g711 support. On Tue, Oct 5, 2010 at 6:16 PM, Mark Holloway m...@markholloway.com wrote: It's the strangest thing. I couldn't get Multicast MoH to work on my BR1 H323 router. I wiped out my call-manager-fallback configuration, re-entered everything, put my router in SRST mode (to practice other things) and just for the hell of it I tried testing Multicast MoH over the PSTN and it worked. I then brought up the Serial interface so the router came out of SRST mode and Multicast MoH is still working as expected. I didn't test Multicast MoH after rebuilding call-manager-fallback but before putting it into SRST. So I'm not sure exactly which one fixed it. However, I did try rebuilding call-manager-fallback a couple of times yesterday and it didn't fix it. My working configuration: call-manager-fallback max-dn 14 max-ephone 2 ip source-address Voice Vlan IP moh music-on-hold.au multicast moh 239.1.1.1 port 16384 route Voice Vlan IP Loopback0 r2(config-subif)#do sh run | sec ccm-m ccm-manager music-on-hold bind Voice Vlan Number On Oct 4, 2010, at 7:30 PM, Prashant Patel wrote: Hi Mark, When you do a show perf query class Cisco MOH device on the server that has the MOH servers registered if you see an increment on the MOHOutOfResources then there is probably a codec mismatch and this increments the counter. The Device Pool assigned to the MOH server needs to have a region that does g711 with all HQ or BR1 or BR2 regions. HTH Prashant On Sun, Oct 3, 2010 at 9:08 PM, Mark Holloway m...@markholloway.com wrote: Sorry James..my mistake. It shouldn't be 10.20.30.254 but rather it should be 192.168.65.254. He it is again (proper) call-manager-fallback max-dn 24 max-ephones 2 ip source address 192.168.65.254 this is the voice vlan default gateway on Vlan302 moh music-on-hold.au piano music file in flash multicast moh 239.1.1.1 port 16384 route voice vlan ip = 192.168.65.254 loop0 ip = 192.1.65.254 ccm-manager music-on-hold bind Vlan302 ip multicast-routing is enabled ip pim dense mode is configured on voice vlan interface and loop0 interface cucm moh audio source and PUB are configured for multicast routing (1 hop) and assigned to mrg br1_mcast_moh which is assigned to mrgl br1 which is assigned to br1 device pool I have a device pool 'MoH' that has a region 'MoH' and is assigned G711 to all other regions. This region is assign to device pool MoH, and device pool MoH is assign to the MoH servers. When HQ calls BR1 and HQ presses hold, the BR1 phone hears piano music. When PSTN calls BR1 and BR1 presses hold, PSTN hears beep beep beep r2# debug ephone moh EPHONE music-on-hold debugging is enabled Oct 4 00:09:50.794: MoH route If Vlan302 ETHERNET 192.168.65.254 via ARP Oct 4 00:09:50.794: MoH route If Loopback0 46 192.1.65.254 via 192.1.65.254 r2#debug ccm-m music-on-hold all Call Manager music-on-hold all debugging is on r2# Oct 4 00:16:08.156: moh_update_rtp: callID 12 dstCallID -1 Oct 4 00:16:08.156: moh_process_ccb: dstadr 192.168.65.30, callid -1, port 21836, codec 16, moh_en 0, moh_addr 0.0.0.0 Oct 4 00:16:08.160: moh_update_rtp: callID 12 dstCallID -1 Oct 4 00:16:08.160: moh_update_rtp: callID 12 dstCallID -1 Oct 4 00:16:08.164: moh_process_ccb: dstadr 192.168.65.30, callid 11, port 21836, codec 16, moh_en 0, moh_addr 0.0.0.0 Oct 4 00:16:08.164: moh_update_rtp: callID 12 dstCallID 11 Oct 4 00:16:08.180: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now connected to 911 N/A Oct 4 00:16:08.180: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now connected to 911 N/A Oct 4 00:16:09.028: moh_update_rtp: callID 12 dstCallID 11 Oct 4 00:16:09.028: moh_update_rtp: callID 12 dstCallID 11 Oct 4 00:16:09.032: moh_update_rtp: callID 12 dstCallID 11 Oct 4 00:16:17.264: %ISDN-6-DISCONNECT: Interface Serial0/0/0:0 disconnected from 911 , call lasted 9 seconds Oct 4 00:16:17.268: moh_update_rtp: callID 12 dstCallID 11 On Oct 3, 2010, at 5:44 PM, James Key wrote: Mark, Looking at your config, a little confused on your ip source address under call-manager fallback and what you have for your route under multicast. One is listed as voice vlan gateway and the other is voice vlan ip, but two different networks. What you have listed for your CUCM config looks correct. Also, do you also have ccm-manager music-on-hold defined on the br1 router? believe this is needed for multicast even though an H323 gateway. James -- *From:* ccie_voice-boun...@onlinestudylist.com [ ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Holloway [ m...@markholloway.com] *Sent:* Sunday, October 03, 2010 7:17 PM *To:* CCIE Voice Maillist *Subject:* [OSL | CCIE_Voice] MoH SRST (Stream from Flash)` I thought I had this
Re: [OSL | CCIE_Voice] SIP Phones in CME
description in voice register pool config mode. On Sat, Oct 2, 2010 at 10:51 AM, CCIE Voice GMAIL givemeccievoice2...@gmail.com wrote: Hi everyone, I am having a hard time remembering what command will affect the number displayed in the upper-right of the phones for CME. With SCCP, I know the description command will effect that number. How do you change this value for SIP phones registered to CME? Thanks for the help, Jeff ___ 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] SIP Phones in CME
Make sure after each change you commit to your SIP phones configuration files you do a create profile under voice reg global. Verify you have your config file ready for download by issueing show voice register tftp and check the mac address of your SIP phone is in the list. A couple of other things to note: You need IP connectivity to your CME router. If your phones are remote to you you need to bind the SIP interface you are sourcing your SIP packets and use that IP address as your source address under voice register global. Authenticate register is a must also if your phones are remote to your SIP CME router. Lastly ensure your voice reg pools have username and password and that you have a number assigned to your phones as the primary number. After all these create profile and if still with troubles please post your configs. Cheers On Sat, Oct 2, 2010 at 12:08 PM, CCIE Voice GMAIL givemeccievoice2...@gmail.com wrote: Hi Dan, Thanks for the response. Before I could test what you had told me, I must have screwed something up. I keep getting a message on both of my CME phones saying “Unprovisioned”. I have reloaded my router and re-configured everything again but I am still getting that message. Has anyone seen this before? Jeff *From:* Daniel Berlinski [mailto:dberlin...@gmail.com] *Sent:* Friday, October 01, 2010 2:55 PM *To:* CCIE Voice GMAIL *Cc:* osl osl *Subject:* Re: [OSL | CCIE_Voice] SIP Phones in CME description in voice register pool config mode. On Sat, Oct 2, 2010 at 10:51 AM, CCIE Voice GMAIL givemeccievoice2...@gmail.com wrote: Hi everyone, I am having a hard time remembering what command will affect the number displayed in the upper-right of the phones for CME. With SCCP, I know the description command will effect that number. How do you change this value for SIP phones registered to CME? Thanks for the help, Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP Phones in CME
I beleive you are missing tftp path flash: under voice register global. Can you try, create profile and let us know? On Sat, Oct 2, 2010 at 1:11 PM, CCIE Voice GMAIL givemeccievoice2...@gmail.com wrote: Hi again, I actually reloaded my router with clean configuration and then re-configured CME, however I am still seeing the same problem. I erased the configurations on the phones before this all happened, so I assume this is maybe part of the problem. I don’t know why it would be though, as the phones are getting IP addresses from DHCP and communicating with CME. This is my relevant configs: - DHCP FOR PHONES - ip dhcp excluded-address 10.5.202.1 ip dhcp pool SC_PHONES network 10.5.202.0 255.255.255.0 option 150 ip 10.5.202.1 default-router 10.5.202.1 - VOICE SERVICE - voice service voip allow-connections sip to sip fax protocol cisco sip bind control source-interface Vlan250 bind media source-interface Vlan250 registrar server expires max 1200 min 500 - VOICE CODEC - voice class codec 1 codec preference 1 g711alaw codec preference 2 g711ulaw codec preference 3 g729r8 - SIP CME CONFIG - voice register global mode cme source-address 10.5.202.1 port 5060 max-dn 20 max-pool 2 load 7945 SIP45.9-0-3S load 7942 SIP42.9-0-3S authenticate register date-format Y/M/D voicemail 4500 url directory http://10.5.202.1/localdirectory create profile sync 0001302544054013 ntp-server 10.5.200.1 mode directedbroadcast - PHONE 1 LINE 1 - voice register dn 1 number 4001 call-forward b2bua busy 4500 call-forward b2bua noan 4500 timeout 10 allow watch name Site C Phone 1 label 4001 - PHONE 2 LINE 1 - voice register dn 2 number 4002 call-forward b2bua busy 4500 call-forward b2bua noan 4500 timeout 10 allow watch name Site C Phone 2 label 4002 - PHONE 1 (7942) - voice register pool 1 id mac 0024.9733.6C28 type 7942 number 1 dn 1 presence call-list dtmf-relay rtp-nte sip-notify voice-class codec 1 username scuser1 password cisco description +442321314001 blf-speed-dial 1 4002 label SCPH2 4002 device privacy off - PHONE 2 (7945) - voice register pool 2 id mac 0024.14B2.F542 type 7945 number 1 dn 2 presence call-list dtmf-relay rtp-nte sip-notify voice-class codec 1 username scuser2 password cisco description +442321314002 blf-speed-dial 1 4001 label SCPH1 4001 device privacy off - TFTP FILES - tftp-server flash:SIP/apps42.9-0-3TH1-22.sbn alias apps42.9-0-3TH1-22.sbn tftp-server flash:apps42.9-0-3TH1-22.sbn alias cnu42.9-0-3TH1-22.sbn tftp-server flash:SIP/cvm42sip.9-0-3TH1-22.sbn alias cvm42sip.9-0-3TH1-22.sbn tftp-server flash:SIP/dsp42.9-0-3TH1-22.sbn alias dsp42.9-0-3TH1-22.sbn tftp-server flash:SIP/jar42sip.9-0-3TH1-22.sbn alias jar42sip.9-0-3TH1-22.sbn tftp-server flash:SIP/SIP42.9-0-3S.loads alias SIP42.9-0-3S.loads tftp-server flash:SIP/term42.default.loads alias term42.default.loads tftp-server flash:SIP/apps45.9-0-3TH1-22.sbn alias apps45.9-0-3TH1-22.sbn tftp-server flash:SIP/cnu45.9-0-3TH1-22.sbn alias cnu45.9-0-3TH1-22.sbn tftp-server flash:SIP/cvm45sip.9-0-3TH1-22.sbn alias cvm45sip.9-0-3TH1-22.sbn tftp-server flash:SIP/dsp45.9-0-3TH1-22.sbn alias dsp45.9-0-3TH1-22.sbn tftp-server flash:SIP/jar45sip.9-0-3TH1-22.sbn alias jar45sip.9-0-3TH1-22.sbn tftp-server flash:SIP/SIP45.9-0-3S.loads alias SIP45.9-0-3S.loads tftp-server flash:SIP/term45.default.loads alias term45.default.loads - PHONE PORTS ---à interface FastEthernet0/2/2 switchport access vlan 150 switchport voice vlan 250 spanning-tree portfast ! interface FastEthernet0/2/3 switchport access vlan 150 switchport voice vlan 250 spanning-tree portfast I am still seeing the phone say unprovisioned. As you suggested I looked at the show voice register tftp command and I can see the SEPmac.cnf.xml statements for the phones. - show voice register tftp - R3#show voice register tftp tftp-server syncinfo.xml url system:/cme/sipphone/syncinfo.xml tftp-server SIPDefault.cnf url system:/cme/sipphone/SIPDefault.cnf tftp-server softkeyDefault_kpml.xml url system:/cme/sipphone/softkeyDefault_kpml.xml tftp-server softkeyDefault.xml url system:/cme/sipphone/softkeyDefault.xml tftp-server SEP002497336C28.cnf.xml url system:/cme/sipphone/SEP002497336C28.cnf.xml tftp-server SEP002414B2F542.cnf.xml url system:/cme/sipphone/SEP002414B2F542.cnf.xml Also, when I look at the status messages on the phone, I see a “Error Verifying Config Info” message. Any help is appreciated, Jeff *From:* Daniel Berlinski [mailto:dberlin...@gmail.com] *Sent:* Friday, October 01, 2010 4:20 PM *To:* CCIE Voice
Re: [OSL | CCIE_Voice] Speed for taking the Lab
HI Pithog Here comes my suggestion: Choose one lab and change the IP addresses in your environment, the number plan, and the physical position of the phones you work with on your desk, introduce infrastructure probs, make sure you make it very hard for your phones to register. Review the troubleshooting IP Tel book on chapter 3 I think speaks about phone registration and there is a white paper on cisco.com that talks about common phone registration probs, make sure you are familiar with those and practice those scenarios so you can see the symptoms happening before you in the stress room chamber On Fri, Oct 1, 2010 at 9:38 AM, Amp amccar...@cciequest.com wrote: Hey Pithog, You ask a tough question my friend. I think some of the things that you need to consider are how well do you know the core technologies and how fast can you correctly configure them? Based upon the forums and the practice labs there are going to be some things that you will need to know how to configure rather swiftly. Will you have CME with SCCP and SIP phones to configure on your lab? Who knows but it would be a good idea to know how to configure CME in a matter of minutes. Can you configure IOS media resources as fast as you can type your name? If not then ask yourself why not. Start configuring H323, MGCP, and Gatekeepers in notepad. If you can do it in notepad with little to no screw-ups then you can do it in the router lightning fast. Are you able to read the question and not over-complicate what's being asked? How fast can you configure COR? Furthermore what's your strategy? Do you plan on configuring once and copying, modifying, and pasting? What do you know really well and what do you need help in? Spend as much time trying to master the areas that you are weak in. Also remember, it is very possible that the IPX labs are more difficult than the actual lab so if you can't get through the IPX labs in less than 8 hours, can you do the core of what's being asked in a timely manner? So in my opinion, configure as much as you can in notepad to ensure you know the configuration steps inside and out. During your steps write out the steps to configure what's being asked. Do this without looking it up and see where you are coming up short. I know I didn't directly answer your question but I hope that helps. Amp Quoting Pithog Oil pithog...@yahoo.com: Please i will like to know if my speed is okay and good enough for the exam, it takes me 8 hours at the moment to finish the ipexpert, Labs, suggestions are welcome on how i can shorthen the time to 4 hours, i really hope its possible, please i need assistance on this. Ultimately i want to know how to manage my time better. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] phone is not taking IP from DHCP server
This is a deep question. Could you tell is what have you done already to troubleshoot this issue? Work your way up from Layer 1 and you will find the issue. There is a good document on cisco.com search for phone registration problems - It is a CUCM 3.x document but very useful. Also, there is the chapter 3 of the troubleshooting book there is great for this. On Tue, Sep 21, 2010 at 1:55 AM, Peterson Gomes pgcristo...@gmail.comwrote: Hello Maybe hub just can see the access vlan and not voice access vlan (auxiliary vlan) ___ 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] Layer 2 Overhead brainstorm
Great point Steve. The archives of this list is in the top 5 most valuable research resources I'm using for my prep. I feel sometimes that most people do not even bother to look at them. Silly mistake because there are loads of great stuff in there! On Thu, Sep 16, 2010 at 2:42 AM, Steve Denney (stdenney) stden...@cisco.com wrote: The “right” value to use is the one that will get you the points in the lab exam. And as Daniel pointed out yesterday, Ben Ng (the lab author) has clearly stated that he uses the values in the QoS SRND. You do have a certain amount of leeway in the calculations (some unknown percentage factor allowed by the graders). Many of the IPexpert materials use what they consider to be a closer to real-world value. Will that value be within the leeway, and get you the points? That’s up to you to decide. Personally – I do wish that the materials were more consistent in the values chosen. As has been stated countless times in the archives - The lab is not a test of best practices or real world scenarios. It’s a test to see how well you can interpret and follow the blueprint and the directions that are given. Draw your own conclusions. :) good luck, sd *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Tam Nhu *Sent:* Wednesday, September 15, 2010 10:31 AM *To:* ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] Layer 2 Overhead brainstorm Still confusing. We know that QoS SRND ues MLP overhead is 13 bytes, but the IPExpert PG always uses 9 bytes. Also, for FRF.12, QoS SRND uses 4 bytes, but the PG uses 6 or 7 bytes, change per lab basic. So what is the right value to use? Thanks, TN. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Layer 2 Overhead brainstorm
Not confuse. Ben Ng himself talks about his preference in this link https://supportforums.cisco.com/message/3010632 As he is the one who designs this exam, I think it is safe to follow his suggestions. On Wed, Sep 15, 2010 at 7:08 AM, ShinGei Yong shingei.y...@gmail.comwrote: Hi all, This is obviously an old question has been repeated N times, but varies varies answer anywhere. According to the QoS SRND, page 1-15 stated: * Multilink PPP (MLP) add 13bytes of layer 2 overhead. * Frame Relay adds 4 bytes of Layer 2 overhead; Frame Relay with FRF.12 adds 8 bytes. My first question is, is this MLP included the the FR overhead as well? So MLP (9 bytes) + FR (4 bytes) = 13 bytes According to Cisco web page: VoIP Per Call Bandwidth consumption, 6 bytes is selected for MLP overhead, which one should follow? Per Call Bandwidth Consumptionhttp://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml Second question is, what is the layer 2 overhead for MLP w/ LFI? According to Cisco End-to-End QoS Network Design book,under chapter 16, the MLP listed here is 10 bytes, + 3 for MLP LFI, total would be 13 bytes for MLP LFI. Confuse? 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
Re: [OSL | CCIE_Voice] Hide a particular user in the Corporate Directory
Are you synced with AD? If you are just remove the last name from AD and the user wont show up. On Sat, Sep 11, 2010 at 12:08 AM, Tam Nhu tamnhu...@gmail.com wrote: Hi Experts, Does anyone know how to hide a particular user, like uccxadmin or crsadmin, in the Corporate Directory so that it does not show up in the Corporate Directory? I searched the OSL and could not find any post about this. Thanks in advance for any suggestions. TN. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] srr-queue Shape vs Share on 3750
Hi Mark I think you need to place a value in that first queue position for the share command line other than zero otherwise IOS gives you an error. That value is ignored because as you have already pointed out, your shape to 25% of the bandwidth is already in place and take precedence on that queue. When you say priority-queue, do you mean simply put Queue 1 or you mean that you believe that you are sizing the priority-queue to 25% of the available interface bandwidth? In your intended config would you add priority-queue out as well? I ask the question because there is no way to size the depth of your priority queue on egress, only on ingress. Cheers On Thu, Sep 9, 2010 at 6:16 PM, Mark Holloway m...@markholloway.com wrote: If I want a priority queue to have 25% of the port bandwidth, I have configured shape 4. I want queues 2, 3, and 4 to share 40%, 40%, and 20% of the remaining bandwidth. All the examples I have seen for shape/share show a value of 1 for priority queue in share regardless of the fact shape is set to 4 (25% of available bandwidth). srr-queue bandwidth shape 4 0 0 0 -- 4 = one 1/4th of total bandwidth; 0 = use Share instead srr-queue bandwidth share 1 40 40 20 -- queue 2, 3, 4 will share 40%, 40%, 20% of bandwidth Why does share's priority queue need a value of 1 if shape is already 4? Is it an indicator saying there is a value for shape so use that instead? Thanks, Mark ___ 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] Fwd: WB 2 LAB 2 Question 5.1 - Gatekeeper Bandwidth Accounting
Bug CSCsl74701 On Tue, Sep 7, 2010 at 1:04 PM, Vccie Vccie voiceccie2...@gmail.com wrote: First off, let me say I have looked for this in the past post's but didn't see anything so I am sorry if this is redundant. But this is the problem I am having. Calling between UCM and UCME works perfect for both Skinny and Sip phones. But the accounting in the Gatekeepr is showing wrong amounts on the calls coming from the UCME to the UCM. But UCM to UCME calls show correct bandwidth accounting (16Kbps). I know the codec used is G729r8 but for some reason it shows g711 bandwidth amounts. Br2 (version 12.4(22)T) dial-peer voice 150 voip destination-pattern [51]... session target ras tech-prefix 1# dtmf-relay h245-alphanumeric rtp-nte no vad Gatekeeper (version 12.4.(20).T4) UCM to UCME = 16Kbps UCME to UCM = 128Kbps After debuging H225 ans1 messages I can see the following admissionRequest - from 3002 to 1002 bandWidth 160 admissionConfirm - bandWidth 160 admissionRequest - from 1002 to 3002 bandWidth 1280 admissionConfirm - bandWidth 1280 infoRequestResponse - bandWidth 160 UCM - 7.01 (H225 Gatekeeper controlled trunk) HQ/BR1/BR2 - each have different Device Pools with G729 intra-device pool Codec. -- So it's the UCM that is responding with a G711 capability's but the call is actually using G729 -- so I am stuck..(and phones show g729 being used) Any help is appropriated. ___ 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] iDivert and + Dialing
Good point Tam The other consideration you may/may not need to look at is that you have visibility of your VM hunt pilot number when you send the call to VM via idivert. Have you checked your css/pt setup that you are providing visibility to the VM hunt pilot number in question? Cheers On Sat, Sep 4, 2010 at 12:14 AM, Tam Nhu tamnhu...@gmail.com wrote: Do you have VM Mask set to in VM Profile? I tested iDivert many times with UC, and it is very straight forward without any issue if the VM Mask set to 4-digit only. TN. Message: 3 Date: Thu, 2 Sep 2010 15:02:13 -0700 From: Cristobal Priego cristobalpri...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] iDivert and + Dialing Message-ID: aanlktikvmk5uojs0drwazkwk=3l0rl4_y1z2jb2cq...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hello All, i was wondering if you know how the iDivert softkey works I'm implementing the + dial on a ucm implementation, the phones can't see each other directly, there is a translation pattern in between that globalizes the number to +1916... and i have a transformation pattern assigned to the CSS that will strip the area code and all of that so the call will be presented to the called party as 4 digits. however if I press the iDivert soft key when a call comes in to a phone the display of the call changes to unknown unknown , the call will never go to voicemail if i press the iDivert soft key again, i see the number in the gobalized format could you please explain how idivert works, in order to resolve this little problem thanks you have a great day ___ 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] Inbound CSS on GK Trunk not working?
Check also if you have a H323 gateway assigned with router HQ. Check the H323 inbound call flow chart in the SRND page 223 in the chapter for trunks for details. You could find yourself in a situation where the css assigned to a H323 gateway takes precedence over the config applied to your trunk 2010/9/1 Roger Källberg roger.kallb...@cygate.se Hi Ryan, Have you verifyed that you don't have a db replication problem? Sincerely *Roger Källberg* CCIE #26199 (Voice) Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ -- *Från:* Ryan Schwab [schwab...@shaw.ca] *Skickat:* den 1 september 2010 06:56 *Till:* 'Ohamien Uhakheme' *Kopia:* 'OSL Group' *Ämne:* Re: [OSL | CCIE_Voice] Inbound CSS on GK Trunk not working? Yep, tried that. Went as far as creating a completely new partition and CSS, and same thing…..no matter what, if a directory number is assigned a partition, it cannot be reached from the GK trunk…. The moment I place the directory number into a NONE partition, with a CSS applied or not to the trunk, it works. I went as far as rebooting my CUCM cluster with no luck…..very odd. *From:* Ohamien Uhakheme [mailto:oham...@gmail.com] *Sent:* August-31-10 10:49 PM *To:* Ryan Schwab *Cc:* OSL Group *Subject:* Re: [OSL | CCIE_Voice] Inbound CSS on GK Trunk not working? Odd.. For SG, try putting 5001 in the None partition, and placing the same CSS on the trunk. Technically that CSS should be able to see the none partition and it should work. If not, then we most likely have something wrong with that CSS... HTH, Ohamien On Wed, Sep 1, 2010 at 12:11 AM, Ryan Schwab schwab...@shaw.ca wrote: Guys, I am trying to route calls from CME to UCM with a Gatekeeper. If I place the DN(5001) on the UCM phone in the NONE partition, the call from the CME (ext 3001) works like it should. As soon as I place 5001 into a partition and configure an inbound CSS on the Gatekeeper trunk, the call from ext 3001 hears the UCM annunciator “Your call can not be completed as dialled”. I am certain the CSS can see the appropriate partition, the trunk has been reset, etc… Is it just getting late here and I’m missing something blatantly obvious?? I should also mention that calls in the reverse direction (5001 - 3001) work with no problems. Anyone have any ideas? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Dial-plan prep work
Hi all How are you guys organizing yourselves in paper prior to tackling your dial plan configs? What I have been doing is xGW - manipulations here for RL/RG combo SITEx - RP [2-9]+6 - RL-SITEx-SITEy yGW - manipulations here for RL/RG combo Dial-p number Translationprofile number I have some pre-defined translation-rules and profiles I use for setting up TON, stripping DNIS incoming, and expanding ANI outgoing. re hardcoded in my head already but I still feel that my approach needs improvement because I always catch myself reading from the paper after my planning stage when configuring the dial-plan. My intent is to only get back to the lab paper when verifying the configuration implemented. How do you guys organize yourselves? Do you guys do it in paper first or rely in your memory? How do you set it up in paper in a scenario where you have H323 gateways in the route-list, dial-peers, translation-profiles and rules for adding + and Type of number? Tks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME Background Issues
Hello What phone are you using? What happens with your debug tftp events? Is the phone looking for the path and files you have uploaded? Sometime you may need to adjust your tftp-server settings with the alias command cheers On Sun, Aug 22, 2010 at 8:01 AM, Cisco CCIE ccieforl...@gmail.com wrote: Yup that's the process I have always followed but end result is always hit and miss. Was wondering who else has ran into similar issues? On 8/21/10, Ashar Siddiqui siddas...@gmail.com wrote: Follow this process and you will never have a miss. At CME router do the following: Ping the tftp server and check connectivity Check if there is a directory on flash like Desktops/320x196x4/List.xml if not then make a directory Create directory in flash “mkdir flash:/Desktops/320x196x4” Copy files across copy tftp://10.10.210.5/List.xml flash:Desktops/320x196x4/List.xml copy tftp://10.10.210.5/small.png flash:Desktops/320x196x4/small.png copy tftp://10.10.210.5/small.png flash:Desktops/320x196x4/small.png Show the path to ephones tftp-server flash:Desktops/320x196x4/List.xml tftp-server flash:Desktops/320x196x4/large.png tftp-server flash:Desktops/320x196x4/small.png Reset ephones Go into Phone Settings à Preferences à Background image and select the new image Open debug tftp events and you will see following on router R3: May 23 15:44:32.367: TFTP: Looking for Desktops/320x196x4/List.xml May 23 15:44:32.367: TFTP: Opened flash:Desktops/320x196x4/List.xml, fd 8, size 152 for process 294 May 23 15:44:32.511: TFTP: Finished flash:Desktops/320x196x4/List.xml, time 00:00:00 for process 294 May 23 15:44:32.907: TFTP: Looking for Desktops/320x196x4/small.png May 23 15:44:32.911: TFTP: Opened flash:Desktops/320x196x4/small.png, fd 8, size 7196 for process 294 May 23 15:44:35.063: TFTP: Finished flash:Desktops/320x196x4/small.png, time 00:00:02 for process 294 May 23 15:44:39.083: TFTP: Looking for Desktops/320x196x4/large.png May 23 15:44:39.087: TFTP: Opened flash:Desktops/320x196x4/large.png, fd 8, size 73628 for process 294 May 23 15:45:00.323: TFTP: Finished flash:Desktops/320x196x4/large.png, time 00:00:21 for process 294 http://tinyurl.com/39cu8eq Ash Cisco CCIE wrote: OK so it appears that this has been happening with others as well. I did a search but none had it resolved. I had this working and then decided to redo the scenario but now it just won't work. Here are all the configurations just incase someone asks for it. Is there a bug with CME that makes this happen? i have never had ANY issues with background images in CUCM but CME is always a hit and a miss. Any help would be HIGHLY appreciated! tftp-server flash:Desktops/320x196x4/List.xml tftp-server flash:Desktops/320x196x4/phonelogoTN.png tftp-server flash:Desktops/320x196x4/phonelogo.png R3#dir Directory of flash:/Desktops/320x196x4/ 88 -rw- 158 Aug 21 2010 16:02:10 +00:00 List.xml 82 -rw- 14567 Aug 21 2010 15:57:08 +00:00 phonelogo.png 89 -rw-3293 Aug 21 2010 15:57:40 +00:00 phonelogoTN.png R3#more List.xml CiscoIPPhoneImageList ImageItem Image=TFTP:Desktops/320x196x4/phonelogoTN.png URL=TFTP:Desktops/320x196x4/phonelogo.png/ /CiscoIPPhoneImageList Aug 21 16:42:32.193: TFTP: Server request for port 49223, socket_id 0x4B6E1B6C for process 351 Aug 21 16:42:32.193: TFTP: read request from host 10.10.202.53(49223) via Vlan400 Aug 21 16:42:32.193: TFTP: Looking for Desktops/320x196x4/List.xml Also I do have the PHONE TYPE under ephones. Thanks in advance! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Sent from my mobile device ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Service URLs
Guys, I'm preparing for situations with different requirements for codecs usage over the WAN and priority queue sizing. I'm using page 134 od CUCM SRND for locating the formulas for calculating voice payload size and packets per second values for supporting me with potential questions involving codecs such as g723, g726, g728, etc. I use this page alongside with QOS SRND page 33 for L2 overhaeads. Part of the formula for calculating the codec payload size in bytes is the codec rate. I'm not able to find a document (searchable in the exam) with the different codec rates. What do you guys use? Thanks On Sat, Aug 21, 2010 at 6:52 AM, Brian Valentine bkvalent...@gmail.comwrote: Thanks for the help with these, Miron and Daniel. I was digging into the CAD install guide for the IPPA service, but the one-button login link is much faster because I don't need Acrobat to get into it, plus I can copy and paste it. Also, I've been getting the IPPM link from the CUPS deployment guide, which is now in wiki format. Not sure how that's handled in the real lab, so the CUCM SRND is much faster and I know I can rely on it being on my candidate desktop PC during the exam (or at least that's what Ben Ng said during the Ask the Expert). I didn't realize that the CUE command line shows the URL needed for Voice View. FYI, it is using the command show voiceview configuration. se-10-10-202-2# show voiceview configuration Phone service URL: http://CUE-hostname/voiceview/common/login.do Enabled: Yes Idle Timeout (minutes): 5 Very nice. I wish all of these were that easy to look up. Brian On Sun, Aug 15, 2010 at 3:05 AM, Miron Kobelski findko...@gmail.com wrote: Hi Brian, these are the quickest methods to get those URLs that I am aware of. I can't check the locations exactly now, as I'm not in the lab, but you should be able to find them: 1) Extension Mobility CUCM Help search for extension mobility checklist 2) IPMA (IP Manager Assistant) CUCM Help search for ipma checklist 3) IPPA (IP Phone Agent) cisco.com UCCX support page configuration examples IPPA one-button login 4) IPPA - One touch login cisco.com UCCX support page configuration examples IPPA one-button login 5) IPPM (IP Phone Messenger) SRND (search PDF for IPPM) 6) VoiceView Express (CUE) go to CLI and run show voicemail voiceview (or similar) or go to GUI Voiceview configuration page (URLs are listed there) hth kobel On Sun, Aug 15, 2010 at 1:43 AM, Brian Valentine bkvalent...@gmail.com wrote: All, I've been trying to improve my speed in general... but specifically in looking up things that I might need in the lab exam. This evening, I've been working on reviewing where to find all the Service URLs. Most are too cryptic to memorize. So... assuming you don't have these memorized, where would you go to look up the following service URLs during the exam? BTW, I have my answers, but want to see what others say to compare with where I found these. Maybe you know of a quicker way to look up one or more of these. 1) Extension Mobility 2) IPMA (IP Manager Assistant) 3) IPPA (IP Phone Agent) 4) IPPA - One touch login 5) IPPM (IP Phone Messenger 6) VoiceView Express (CUE) Secondary question: Am I missing any? Are there any other IP Phone Services that would be fair game in the lab exam? The only other one I can think of off hand is the VoiceView for CUC, but that requires another server. Does anyone think it could be considered a testable topic? Brian Valentine ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] QoS bandwidth calculations
As a suggestion I would check for the overheads sizing at page 33 of QoS SRND and for the formulas I would check UCM7 SRND page 134 HTH Daniel On Sun, Aug 15, 2010 at 1:44 AM, Stutz, Bernhard st...@pandacom.de wrote: Hi, that's what i found at proctorlabs solutions. That's why i am asking, i want to know for sure how to calculate these values. cheers, Bernhard -- *Von:* Trying 2nd CCIE [mailto:dukelon...@gmail.com] *Gesendet:* Sa 14.08.2010 15:41 *An:* Stutz, Bernhard *Cc:* OSL Group *Betreff:* Re: [OSL | CCIE_Voice] QoS bandwidth calculations Hi, What about the default 10ms sampling rate? What is the calculation formula? Thanks and Regards, John On 14 August 2010 21:36, Stutz, Bernhard st...@pandacom.de wrote: Hi, Has somebody found an overview of actual bandwidth consumption including L2 overhead per packet? What i know so far is following: all at 20ms sampling rate When FRF.12 LFI and RTP: L2 (FR) = 7 bytes IP/UDP/RTP = 40 bytes ilbc codec =38 bytes When FRF.12 LFI and cRTP: L2(FR) = 7bytes IP/UDP/RTP = 2bytes G729 codec = 20 bytes when MLP LFI and cRTP: L2(FR) = 9bytes IP/UDP/RTP = 2bytes G729 codec = 20 bytes cheers, Bernhard ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Infrastructure Brainstorm request
Hello List I would like to initiate a thread which I invite all of you to collaborate sharing ideas on the following: What would you think could break DHCP relay traffiic to succeed with the scenario below: (CUCM DHCP) 3750WAN-RTR WAN WAN-RTR Etherswitch IP Phone So far I can think of the following: 1- CSA enabled in CUCMs 2- Vlan access-maps in Switch 3- DHCP snooping configured in switch 4- service dhcp disabled in the branch router 5- L2/L3 problem within the network 6- ACLs in L3 interfaces along the way 7-DHCP service disabled in CUCM 8-DHCP service server/subnet page misconfiguration I'm trying to go through all sections of the blueprint I believe I completed 100% at this stage and only practice troubleshooting scenarios. Would be thankful for any input Cheers ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Calling Number Display on Phone - Calls to PSTN-WAN
Hi Matt Did you get this going? I tried by doing pretty much everything I can thing of: CdPTP at trunk Manipulations at route-pattern, translation-pattern, dial-peer of cube and num-exp. no supp service command under voice service voip and playing with the connected party settings. I think there is no way to stop the cld updates coming from the backbone gateway. Did you find any different? Cheers On Thu, Aug 12, 2010 at 12:29 AM, Ashar Siddiqui siddas...@gmail.comwrote: What is your gatekeeper config? What prefix you matching at GK? Ash -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Matthew Berry Sent: 11 August 2010 05:21 To: CCIE Voice OSL Subject: [OSL | CCIE_Voice] Calling Number Display on Phone - Calls to PSTN-WAN All - I'm sending calls to 9011.91!# across the local gatekeeper to the PSTN-WAN backbone gatekeeper. What I notice is that no matter what I try to do in order to manipulate the calling number displayed on the IP phone, it always shows 6745738932. Meaning, if I dial 9011-91-67-4573-8932 on my Cisco IP phone, the display shows up as 6745738932. I am telling the system to stripp the 9011, but where does the 91 go? I was told by a friend that this is due to the num-exp command used on the PSTN-WAN backbone gatekeeper. Is that true? I'm concerned that this may bite me in the lab. The no supplementary-service h225-notify cid-update command does not fix this. Ideas? Thanks, Matthew Berry ciscovoiceg...@gmail.com http://ciscovoiceguru.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Remote Gatekeeper PSTN Config
Edwin what does your debug gatekeeper main 10 show? Do you have IP connectivity between them? This should not be this hard as LRQ routing decision when configured correctly works in the very early stages of the ARQ flow chart SRND page 520 is very helpful to understand this. On Thu, Aug 12, 2010 at 8:44 AM, Edwin Dotson edot...@ams.net wrote: I am trying to setup a remote gatekeeper on my PSTN Router and cannot get the Remote and Local gatekeeper to talk. Does anybody have a sample config for the PSTN Gatekeeper? Thanks, *Edwin Dotson* Senior Systems Engineer CCNA, CCVP Cisco Unity Support and IP Contact Center Express Specialist *AMS.NET* 925-245-6144 – Office 925-960-6644 – Fax www.ams.net *New Content Information* Events, Webinars, Videos, Case Studies, Downloads More! Click Here http://www.ams.net/ *Cisco Award Winner *Vertical Partner of the Year, Voice Partner of the Year… Read More http://www.ams.net/company/News.asp ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Feedback on top documentation sources for understanding core topics of the exam
Hello List I'm requesting feedback from all of you on what is the top documents or chapters of books you found most useful for your preparation. I know this is a far to disperse topic but I am interested to know which documents you found the top ones. I will provide and example below: In my case I found chapter 10 of the Troubleshooting book extremely useful to understand call preservation. In the support page section on troubleshooting configuration Technotes, which will be available to be viewed during the lab (I hope) I found the document understanding cisco IOS gatekeeper call routing a very succinct and source of info for h323 stuff. In fact that whole section is a real cool source of info and I recommend all of you to read as many technotes as possible before you go in that lab room. Haven't attempted the voice lab myself but in the past in other tracks the technotes proved to be a very clever way to prepare for the challenges that will be posed in the exam. On the IP phone registration front the troubleshooting technote Troubleshooting Cisco IP Phone Registration Problems with Cisco CallManager 3.x and 4.xhttp://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a008009485a.shtmlis very nice too. What other documents have you found enlightening? Anything on the SIP front? If you are keen to reply please be as much specific as possible. Cheers Daniel ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MGCP failover and call preservation
Hi Matthew I'm going to test this to check if I can reproduce your fault but a few questions first: Can you confirm to us that your gateway re-registers successfully to the backup CUCM? After registering you should see and AUEP and AUCX coming from CUCM interrogating the calls the gateway preserved. Are you using debug mgcp packets to view these? So, if this is not happening maybe you have got something blocking communication there? How are you testing your failover? By stopping the service in CUCM or by a null route/ACL? Cheers On Mon, Aug 9, 2010 at 3:08 PM, Matthew Hall 1.matt.h...@gmail.com wrote: Not the question you might be thinking. It seems really basic, but I must be forgetting something. When I preserve a call to the PSTN, once mgcp has lost connectivity to the primary call manager, If I hang up the voip phone, the PSTN line does not hang up. I don't see UCM send anything to the MGCP gateway in my debugs and consequently a disconnect never get sent to the PRI. The call eventually disconnects due to timeout. This occurs whether the primary call agent comes back online or not. What am I missing? ___ 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] LAN QoS Priority and buffer size
Thanks very much for this. I will read more carefully about this topic looking carefully for the words in the documentation I could not find to come with some conclusion as yours. Your answer makes sense and it is very objective and I appreaciate that because this topic has beend discussed many times here but with very loose ideas. Do you mind sharing with us what path led you to this conclusion? Was it a document you read that explicitly said that the buffers are used to store the excess traffic and not to provide the physical pool of memory allocation to be used to each queue? Again I thank you for your objective and illustrative help! Daniel On Wed, Aug 4, 2010 at 1:59 AM, Wafik Maher wafikma...@gmail.com wrote: Hi Daniel, I absolutely agree with you on the first part, regarding that enabling the priority (expedite) queue will exclude queue 1 from the SRR shaping and sharing. However, I don’t think that it is possible to control the bandwidth percentages using the buffer size allocation “mls qos queue-set output 1 buffers 10 10 26 54”, simply because of the fact that the buffer is used to store the excess traffic when the input rate is higher than the output rate of a certain traffic. To illustrate my point let me introduce an example 3750 fastethernet (100Mb) 10 % are allocated to priority traffic 50 Mb Average Total Input rate of priority traffic On the above example, at the output the priority traffic can (theoretically) go up to full link speed 100 Mb, so apparently the input rate is not exceeding the output rate and the average output rate would be equal to the input which is 50 Mb (50 % not just the 10 % of the buffer size). As a matter of fact the buffer in this case is not expected even to fill the 10 %, it would just fill a small percent to accommodate with traffic spikes and the small processing delay. ___ 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] class-based shaping
Hello there Check this path in the SRND WAN Aggregator QoS Design - WAN Edge Link-Specific QoS Design - Frame Relay - Slow-Speed (£ 768 kbps) Frame Relay Links, then turn the page and you will find it. HTH On Wed, Aug 4, 2010 at 5:12 AM, Stutz, Bernhard st...@pandacom.de wrote: Hi all, In Proctorlabs there is always mentioned that the QOS SRND provides a adequate example for class-based shaping but i can't find them. Also i heard rumors that the QOS SRND document provided at the labs is not at Cisco Website available anymore. However i can find a version here: http://www.cisco.com/en/US/docs/solutions/Enterprise/WAN_and_MAN/QoS_SRND/QoS-SRND-Book.html but i can't find there an example for class-based shaping. Does anyone know where i can find such a adequate example? good luck and have fun in your studies, Bernhard ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] presence call-lists ?????
hello there I do not think you can enable presence call-lists on a per device basis. By looking at your requirements I believe it is possible to configure a phone to view presence call-list information of selected extensions by enabling allow watch for those desired extensions you wish to monitor. Eventhough you have enabled presence call-list globally you can still be selective with which DNs you want to view updates with allow watch I believe that the per-device presence call-list is a feature for 7.1 onwards. People please correct me if I'm wrong. On Sun, Aug 1, 2010 at 3:01 PM, voiceie2b 2xcci...@gmail.com wrote: Is it possible for ONLY 1 phone to be able to view presence information of another phone in its local directory ? The only way I can get the presence call-list to work is when I enable the command globally. If i do not enable globally and enable it just under 1 phone it does not work. ! presence NO presence call-list ! ephone 1 presence call-list ! ephone-dn 2 allow watch ! Is it possible to get presence call-list to work without enabling globally and to just enable it locally on the phone ? ___ 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] SRST dial-peer behaviour
Why aren't you sending all digits to the IOS gateway and doing your manipulations there? It seems to me a simpler solution to have. Your dial-peer 7 is matching the dialed numbers first because dial-peer matching behaves like that- just like a route-pattern with urgent priority checked. as soon as it matches it sends the call -. You could remove $ from dial-peer 7 and put a T in the end of it to get the matching process to wait a bit but I personally find it better to send all digits from CUCM and have all manipulations done there. If you need to change your calling device display as well then you can try by just sending the digits as they hit the outbound dial-peer without any manipulations there because the 9 will get stripped anyway as it is an explicit match. If that does not tweak the caller display then plug a translation-profile or num-expansion then if that does not work either you could always do a predot in route-pattern and put the 9 back on the route list as last resort. I will try this as soon as I can. It sounds like a hot topic this one! On Thu, Jul 29, 2010 at 4:39 PM, Erwan Erwan e_er...@yahoo.com wrote: Hi Experts, I am trying to configure so that *calling phone* will show 7 digit *To 8884343* in SRST and Normal mode. I use dial-peer 7 pots in normal mode (send 8884343 from Route Pattern to BR-1 GW) RP Local : 9.[2-9]xx , predot , send to BR-1 (H323) , hit dial-peer 7 pots And it did show 7 digit 8884343 in my Calling phone BR-1 dialpeer -- dial-peer voice 7 pots destination-pattern [2-9]..$ port 0/0/0:23 forward-digits all However when I dial 98884343 in SRST mode,I expect it will use dial-peer 9 pots (because I have to dial 9 for local call) dial-peer voice 9 pots destination-pattern 9[2-9]..$ port 0/0/0:23 forward-digits 7 But the call from phone always hit dial-peer 7. And if I shut down dial peer 7, local call will work fine in SRST. But why it hit dial-peer 7 in SRST for 98884343 , which I think dial-peer 9 is more precise match ?? And if I tested using csim start 98884343 in SRST , it will hit dial-peer 9 (which is right for this case). But if from IP phone it will use dial-peer 7 when I dial 98884343 in SRST mode. Anybody know why and shade light on this ? Thks ___ 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] SRST dial-peer behaviour
I hate to post things without having access to equipment to try it myself as it spams everyone's mailboxes. I acknowledge that and apologize for that to everyone. I wont have access to anything until tomorrow and the temptation to reply is stronger than me. Are you an IP expert customer? Did you watch Vik's classes? He teached that the cosmetic effect on your calling device when you place a call through a H323 gateway is achieved by manipulating anywhere as per list below: RPRLRGIn peerNum-Exp of h323 gateway. Any manipulation done at the outbound dial-peer will not trigger the cosmetic effect you are after. I am still of the opinion that you are complicating things. If I was in your shoes I would do the following: 1- Send the 9+7 digits to CME and don't do any manipulation in CUCM at all. Then you could try these different options that I will try as soon as I get my hands on my rack! a) translation-profile on inbound voip dial-peer to strip the 9 b) num-expansion globally configured in h323 gateway to strip the 9 c) do not do any manipulation at all on the outbound dial-peer with destination-p 9[2-9]+7 as the 9 will get stripped for you anyway On Fri, Jul 30, 2010 at 3:52 AM, Erwan Erwan e_er...@yahoo.com wrote: hi Daniel, tks, here is the reason: 1. If I send all digit, with 9 (Phone always display To 98884343) What I want is 8884343 2. put T at the end of dial-peer 7, i tried to avoid that , as it will capture my other call. I like to use precise one for each call. 3. do a predot in route-pattern and put the 9 back on the route list as last resort === this is work and I used it , however you can not do it for RL-Standard , as it will affect other RP which use RL-Standard Any idea how to achieve this? --- On *Thu, 7/29/10, Daniel Berlinski dberlin...@gmail.com* wrote: From: Daniel Berlinski dberlin...@gmail.com Subject: Re: [OSL | CCIE_Voice] SRST dial-peer behaviour To: Erwan Erwan e_er...@yahoo.com Cc: ccie_voice@onlinestudylist.com Date: Thursday, July 29, 2010, 3:02 PM Why aren't you sending all digits to the IOS gateway and doing your manipulations there? It seems to me a simpler solution to have. Your dial-peer 7 is matching the dialed numbers first because dial-peer matching behaves like that- just like a route-pattern with urgent priority checked. as soon as it matches it sends the call -. You could remove $ from dial-peer 7 and put a T in the end of it to get the matching process to wait a bit but I personally find it better to send all digits from CUCM and have all manipulations done there. If you need to change your calling device display as well then you can try by just sending the digits as they hit the outbound dial-peer without any manipulations there because the 9 will get stripped anyway as it is an explicit match. If that does not tweak the caller display then plug a translation-profile or num-expansion then if that does not work either you could always do a predot in route-pattern and put the 9 back on the route list as last resort. I will try this as soon as I can. It sounds like a hot topic this one! On Thu, Jul 29, 2010 at 4:39 PM, Erwan Erwan e_er...@yahoo.comhttp://us.mc1205.mail.yahoo.com/mc/compose?to=e_er...@yahoo.com wrote: Hi Experts, I am trying to configure so that *calling phone* will show 7 digit *To 8884343* in SRST and Normal mode. I use dial-peer 7 pots in normal mode (send 8884343 from Route Pattern to BR-1 GW) RP Local : 9.[2-9]xx , predot , send to BR-1 (H323) , hit dial-peer 7 pots And it did show 7 digit 8884343 in my Calling phone BR-1 dialpeer -- dial-peer voice 7 pots destination-pattern [2-9]..$ port 0/0/0:23 forward-digits all However when I dial 98884343 in SRST mode,I expect it will use dial-peer 9 pots (because I have to dial 9 for local call) dial-peer voice 9 pots destination-pattern 9[2-9]..$ port 0/0/0:23 forward-digits 7 But the call from phone always hit dial-peer 7. And if I shut down dial peer 7, local call will work fine in SRST. But why it hit dial-peer 7 in SRST for 98884343 , which I think dial-peer 9 is more precise match ?? And if I tested using csim start 98884343 in SRST , it will hit dial-peer 9 (which is right for this case). But if from IP phone it will use dial-peer 7 when I dial 98884343 in SRST mode. Anybody know why and shade light on this ? Thks ___ 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] First attempt
I've heard this from Cisco employees as well that when the proctors reach the conclusion that the candidate cannot make it up the 80% they just stop the correction. This is something I will definetely ask whenever there is another ask the expert forum. By the way has anyone ever looked for this info in the ask the expert archives? I know this forum is packed with Cisco staff. Can any of you clarify this for us? On Fri, Jul 30, 2010 at 7:29 AM, CCIE Voice GMAIL givemeccievoice2...@gmail.com wrote: I have heard this from a couple of people and even on this mailer. That is why I am bringing it up. I am not 100% sure if it is accurate or not. *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Graham Hopkins *Sent:* Thursday, July 29, 2010 12:09 PM *To:* OSL Group *Subject:* Re: [OSL | CCIE_Voice] First attempt I don't see how this can be correct, if it is it makes the report meaningless. You could screw up a few early sections, fail on 79% and still have most of the report as 0. Of course as the score report is subject to NDA we'll never know. Still Ohamien keep working on it and you will get there. Graham On 29 Jul 2010, at 19:59, CCIE Voice GMAIL wrote: It’s also important to note, and correct me if I’m wrong, that the 0’s don’t necessarily mean you configured that section incorrectly. To my knowledge, once you lose more than 20 points, they simply stop grading your exam. So the later section may have 0’s but you configured them correctly. I feel like this is a big problem with the already vague score reports. I wish they would change this. If you are paying $1400, you deserve a full report in my opinion. *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Ashar Siddiqui *Sent:* Thursday, July 29, 2010 11:25 AM *To:* Ohamien Uhakheme *Cc:* OSL Group *Subject:* Re: [OSL | CCIE_Voice] First attempt I am sure you will figure out what mistakes you made which resulted in 0%. I know its very hard to find out when you are sure your solution is 100% but believe me I have been through this and you will come to know how a tiny mistake in that particular section or may be in some other section resulted in 0% for this section :) I hope you pass in 2nd attempt. Don't forget to break down your scores and analyze exactly which question you lost points. That will help you to work out on specific areas. Ash Ohamien Uhakheme wrote: Hey guys -- I've been lurking for a while, so I figured that I'd chime in. I sat for my first attempt yesterday with less than passing results. Like other people have mentioned, it is heart breaking to see 0% in areas that you are sure that you nailed completely. It's cool though, I needed to get the psychological first attempt out of the way, and I will probably schedule again for early September. IPExpert is spot on with their training material, and I definitely appreciate the effort that has gone into it. Thanks guys, Ohamien -- ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Layer 2 QOS
In my opinion this is done by adjusting the buffer size for queue 1 and applying it to a queue-set. srr shape statement in my opinion means nothing in relation to adjusting priority queue size. http://onlinestudylist.com/archives/ccie_voice/2010-July/069398.html On Fri, Jul 30, 2010 at 1:19 PM, Jeff Cotter jcot...@voxns.com wrote: How would you enable the priority queue AND make sure queue 1 has 10% of the bandwidth. The documentation states that if the priority queue in enabled, shape and share configuration for that queue is ignored. So how do you accomplish this without using Shape command. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CBarge in SRST mode
Guys Are you testing this particular feature in preparation for the lab exam or for your work/production requirements? I ask the question because the version of IOS all lab routers are running with is 12.4(20)T2 as per Ben Ng during the last ask the expert forum. On Tue, Jul 27, 2010 at 4:04 AM, Graham Hopkins ghopk...@wolf-rock.co.ukwrote: On my own kit 2801/2811 12.4.24Txx ( will check exact version when I get home). Yes I can only get it to work with auto provision none if I use privacy off on the ephone, it then appears to take the phone template that refers to the remote in use soft keys but no privacy button appears on the phones. I tend to agree that this must be IOS related as everyone gets slightly different results. Just wanted to explore all the options in case a lab question asked not to configure the ephones and was also thinking about the comment on the IP Expert blog - from Ben Ng I think - saying that there are bugs and we ought to know the workarounds Graham On 26 Jul 2010, at 16:44, Mark Holloway m...@markholloway.com wrote: Graham, Are you configuring this in your own lab or using Proctor Labs? I am using my own lab (2800's, 12.4.24T3, 7965 phones) and I couldn't get cBarge to work in SRST with auto provision none. Others using Proctor Labs said they could get it to work. Perhaps it's a difference between IOS versions and/or phone types. I literally tried everything. On Jul 26, 2010, at 6:59 AM, Graham Hopkins wrote: Been following the thread on this and have concerns about the ephone-template not appearing to work. The only but I can find that relates to this is CSCsx15347 which refers to a G.729 codec in the ephone -template not being used until after a reboot. The only way I can get this to work without specifying privacy off under the ephone is to run with srst mode auto-provision all and then save the config and reboot - the ephone-template then works privacy button as well . Config below. Anyone have any further thoughts on how to do this without using auto-provision all. Anyone found a way to do it with auto provision none and the ephone template - no manual configuration of the ephone? telephony-service sdspfarm units 4 sdspfarm tag 1 br1-conf no privacy conference hardware srst mode auto-provision all srst ephone template 1 srst dn line-mode octo max-ephones 4 max-dn 8 ip source-address 10.10.201.1 port 2000 system message CCIE SRST Fallback voicemail 912123945600 max-conferences 8 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Jul 21 2010 11:48:33 ephone-template 1 privacy off privacy-button softkeys remote-in-use Newcall CBarge ephone 1 mac-address 0026.CB3D.2888 ephone-template 1 button 1:1 2:2 3:3 ! ! ! ephone 2 mac-address 0021.D8B8.EDDF ephone-template 1 button 1:4 2:3 ! Regards Graham ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Called Party display
Ì have written down on my notes from the VoDs I watched and from testing done that H323 gateways and cosmetic effect in calling device display is achieved by the following on calls that traverse H323 gateways: Any digit manipulation performed in: RP or RL or RG or Inbound dial-peer or Num-Exp will affect the calling device display whereas manipulation performed in outbouond dial-peer does not affect the calling device screen. I think this explains your workaround and to be honest I think the for you to achieve your requirement you got to go down this track. The other option you may have is to do called party xfoms on the egress BR1 gateway and then use the no supplementary-service h225-notify CID-update on the H323 gateway or do your H323 DNIS dgt manipulation on the outbound dial-peer. I'm unware of any service params for this but sometimes when I am 100% sure something is correctly configured but does not work I reboot my servers. Have you given your boxes a kick after attempting the no supplementary-service h225-notify CID-update config? I need to try this myself but wanted to provide a suggestion. let us know what your findings are. I'm still a bit far from the lab you are doing. Cheers Daniel On Sat, Jul 24, 2010 at 6:21 AM, Brian Valentine bkvalent...@gmail.comwrote: no supplementary-service h225-notify cid-update doesn't seem to help. I had an epiphany and figured out a work around to accomplish the task. What the PG suggests seems to work fine, but only on an MGCP gateway. I had to build an additional dial-peer in my BR1 gw with destination-pattern 415888 (forward digits 7). So, from CUCMs perspective, it sends the gateway 4158884343. If I do the manipulation on the H323 gateway, it works. HQ Phone 2 will basically send whatever CUCM sends an H323 gateway. Maybe there is a service param somewhere? Brian On Fri, Jul 23, 2010 at 2:15 PM, Matthew Berry ciscovoiceg...@gmail.com wrote: Voice service voip No supplementary-service h225-notify CID-update Matthew Berry **Sent from my iPhone** Skype/Twitter: ciscovoiceguru Google Voice: +1 612 424 5044 On Jul 23, 2010, at 12:31, Brian Valentine bkvalent...@gmail.com wrote: I'm working on Vol2 Lab7 Task2.4. The task involves the following: HQ phone 2 dials 914158884343. Prefer to use TEHO to route the call out BR1. Local telco expects 7 digits. BR1 is an H323 gateway, so CUCM sends it 98884343. The gateway strips the 9 before sending to telco. Second choice gateway is the HQ gateway, which is MGCP. Local telco will expect 11 digits. CUCM would send the gateway 14158884343. Regardless of which gateway the call goes out the HQ Phone 2 display should say: To 4158884343. Got the call routing and redundancy down fine. That's works well enough. The problem is that no matter what I do, it seems to convert the display on HQ Phone 2 to match whatever digit manipulation was required by the egress gateway. The proctor guide says: The display on the Calling phone will be derived from the Route Pattern manipulation although the actual digits the UCM sends to the gateway is determined by the Route List/Route Group Called # transformations. So, I tried that. I tried doing all my digit manipulation on the RL details level and use the XX as the Called Party transformation on the Route Pattern level. Call goes through, but the HQ Phone 2 still displays To: 98884343. Next I tried setting the RL details to leave it as 415888 and used a Called Party Transformation Pattern at the gateway level to convert the call. I got the same result. Call succeeds. The display on HQ Phone 2 shows To: 98884343. What am I missing? Is this task possible? Thanks in advance, Brian ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] LAN QoS Priority and buffer size
Hello Can someone confirm my understanding. The below question implies the use of priority-queue out inteface command. For adjusting how much bandwidth is given to the egress priority queue of a 3750/3560/2960 switch the interface command: srr-queue bandwidth shape *means nothing* The srr command that tunes the buffer size of memory to be given to queue 1 is the one that will adjust the priority-queue depth required. Feedback please ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] LAN QoS Priority and buffer size
Thanks sir for your reply. This is my understanding for quite a while and i hope i am not wrong for all this time. So i'm happy that you agree. This goes inline to the document 3750 QoS configuration examples - The document does not state this objectively and I'm finding weird the fact of seeing lots of posts lately in this forum of people adjusting priority queue size by tweaking the wrong place so keen to hear any contrary intelligent opinions out there. If anyone out there disagrees please join the discussion and don't be shy. On Sat, Jul 24, 2010 at 12:49 PM, Randall Saborio ill2...@gmail.com wrote: Hi Daniel, I had to review again my notes and the documentation to tell for sure (wish I knew it out of my head as earlier I was studying a lot of the lan qos theory). You are correct, the srr-queue bandwidth shape means nothing when you configure the priority queue out. As it says on the doc: All four queues participate in the SRR unless the expedite queue is enabled, in which case the first bandwidth weight is ignored and is not used in the ratio calculation. *The expedite queue is a priority queue, and it is serviced until empty* before the other queues are serviced. You enable the expedite queue by using the priority-queue out interface configuration command. So what I get is the settings are ignore completely for the calculation of shared bandwidth for the other queues, and because the queue is serviced until empty. I'm all dizzy today from studying the LAN QoS and still can't say I know it all. :-/ On Fri, Jul 23, 2010 at 5:49 PM, Daniel Berlinski dberlin...@gmail.com wrote: Hello Can someone confirm my understanding. The below question implies the use of priority-queue out inteface command. For adjusting how much bandwidth is given to the egress priority queue of a 3750/3560/2960 switch the interface command: srr-queue bandwidth shape means nothing The srr command that tunes the buffer size of memory to be given to queue 1 is the one that will adjust the priority-queue depth required. Feedback please ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Randall da ill Saborio CCIE Voice Wannabe #10054675811 (Real number coming this July 2010) ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Outcall and CUE in CCM mode
Hello On CUE if I have a GDM with one of its members br2ph2-3002, how do I get MWI working to lamp the member's phone? I have unsolicited notify configured as MWI method. Tried to enable outcalling and do the trick with the num-exp but outcall is not supported on cue with CCM mode. Suggestions please. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Outcall and CUE in CCM mode
Hello List Apologies for my previous incomplete post. This environment has the BR2 router in CME SRST mode. The CUE is licensed for CCM integration. While in CME SRST mode I'm trying to get the BR2-PH2 line 1 that is 3002 to have its MWI to light when the GDM receives a message. BR2PH2 is voicemail subscriber to CUE and is a member of the GDM in question. BR2PH2 is a SCCP phone Thanks On Sat, Jul 24, 2010 at 1:18 PM, Daniel Berlinski dberlin...@gmail.comwrote: Hello On CUE if I have a GDM with one of its members br2ph2-3002, how do I get MWI working to lamp the member's phone? I have unsolicited notify configured as MWI method. Tried to enable outcalling and do the trick with the num-exp but outcall is not supported on cue with CCM mode. Suggestions please. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol2 Lab8 Q4.7 Broadcast Messages
What happens when you call from br2ph2? You need your dial-peer for CUE to also match 3300 and then if my memory does not fail now, you should hear an option for bcast a message or something similar. If I recall correctly I called from br2ph2 so no dn was created for re-direction. Let me know how you go. On Thu, Jul 15, 2010 at 3:00 AM, Kevin Damisch kevin.dami...@vitalsite.comwrote: What am I missing on this one? As shown in the PG, I have added br2ph2 as an owner/member of the Broadcasters group and assigned extension 3300 to it? What else is needed? DN or dial peer? Using a dial peer gives me a busy signal. Using a DN forwarded to CUE give me “there is no mailbox associated with this extension. Or, is this accessed via VoiceView? Was this left out of the PG or is this one of those “you should be smart enough to figure it out”? J -- This communication (including any attachments) is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If you are not the intended recipient, any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify Vital Support Systems at 515 334 5700 and delete or destroy all copies and the original document. ___ 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] VATS in the VOD
Hi Brian This is a very good point. My take on this is that the fragment size should be based upon the MINCIR value (adaptive value) because that is the resulting shaped rate when packets are seen in the priority queue. In other words when there is traffic in LLQ there is a reduction of the sending rate from CIR to minCIR and only then VATS will activate FRF.12 end-to-end fragmentation. In other words there is no fragmentation when there is no packets in the priority-queue and there is no fragmentation when traffic is being sent at CIR rate. The IOS documentation @ http://www.cisco.com/en/US/docs/ios/wan/configuration/guide/wan_fr_vats_frag_ps6441_TSD_Products_Configuration_Guide_Chapter.html Frame Relay Voice-Adaptive Fragmentation Frame Relay voice-adaptive fragmentation enables a router to fragment large data packets whenever packets (usually voice) are detected in the low latency queueing priority queue or H.323 call setup signaling packets are present. When there are no packets in the priority queue for a configured period of time and signaling packets are not present, fragmentation is stopped. So I think it makes little sense to me (please if someone disagrees - let me know as I need to confirm my understanding too) to base the fragment size at the rate that is used when there is no fragmentation. By the way, this goes inline with Volume 2 Lab 7 QoS section which I hope its solution is correctly outlined. Cheers On Tue, Jul 6, 2010 at 7:28 AM, Brian Valentine bkvalent...@gmail.comwrote: Vik, Well done on the VoD product. It's really very helpful. I was going through the WAN QoS video today. Question for you on VATS - should the fragment size be based on the adaptive rate or the cir? In the VoD you mention that the fragment size in the class should be 960, not 80. I understand that 80 was nonsense, but I was thinking it should be 480. I would appreciate it if you could clarify for me. See attached screenshot. Thanks in advance! Brian Valentine ___ 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] Music on Hold
Hello Afzal From what you told us it appears that you need to adjust your ServerMax Hops to a value greater than 1 in order for the Mcast stream to reach your branch phones. have you tried doing that? On Tue, Jun 29, 2010 at 7:05 AM, Afzal Bhutta azhar.bhu...@gmail.comwrote: Hello, Here is some more details, MOH is multicast on 239.1.1.1 port 16384.Allow multicasting is enable on CUCM-PUB. CallManager MoH Server Increment Multicast on = IP Address CallManager MoH ServerMax Hops = 1 MOH Audio Source: SampleAudioSource (1) = Allow Multicasting In Media Resource Group = Use Multicast for MOH Audio (This is enable) CME is completely separate side,It is not participating in this Scenario. IP Voice Media Streaming App is enabled for G729 and G722 in service parameter.(Cisco IP Voice Media Streaming App = 711 uulaw and 729 Annex A selected) I have MOH region with G711ulaw enable with all other region with codec G711ulaw. HQ device pool using MRGL SiteB device pool using MRGL MRGL contains MOH-PUB-MULTI-RG All phones within site (Intra-site) using G711ulaw where as between site (Inter-site) they are using G729ulaw. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Lab 5 Volume 2 LLQ sizing and RSVP CAC
Hello list Volume 2 lab 5 has a scenario asking us to allow for 4 concurrent g729 calls over Frame FRF.12 LFI using RSVP for CAC. Proctor Guide has calculated the size of the priority queue without taking into account that first call prior to capabilities exchange that RSVP negotiates at 40Kbps. In addition Proctor Guide has used Frame Relay payload of 4 Bytes instead of 8 Bytes for FR with LFI. I answered this question as follows: For 4 g729r8 concurrent calls over the WAN using RSVP for CAC: compressed ip/udp/rtp=2bytes FRF.12=8Bytes g729 payload @ 20ms=20Bytes 30*50*8/1000=12Kbps per call so 3 calls=36Kbps 1 call all @ worse case scenario compressed ip/udp/rtp=2bytes FRF.12=8Bytes g729 payload @ 10ms=10bytes 20*100*8/1000 = 1 call 16Kbps So 4 calls=36kbps + 16Kbps= 52Kbps configured in priority queue Can anyone let me know if my approach is right or wrong and if wrong why? Thanks a lot Daniel ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 5 Volume 2 LLQ sizing and RSVP CAC
Thanks Kobel for your explanation. It does make sense to me that small voice packets do not grow in size to the point of being fragmented. The only thing that I'm not too sure is whether or not this would be something that Cisco would be expecting to see as a valid answer if such a question was asked in the exam. I guess I would ask the proctor because the lab exam is usually far from reality. Hello Mouhammad I disagree with your statement Finally, I know that both LLQ and ip rsvp bandwith values must be identical and calculated as = (N-1) calls at 20 mSec + 1 call at 10 mSec Why would you always match those two values? Are you calculating these with or without layer 2 overhead? There is an example in the UCM 7 SRND page 3-64 which describes RSVP calculation examples without taking any layer 2 overhad into account. There is a note on page 3-64 that states Unified CM does not include SRTP overhead or the L2 overhead int he RSVP reservation. and then it says that the layer 3 IP rsvp bw statement must take into account any SRTP traffic and the L2 priority queue must also be over-provisioned if SRTP is present. How do you guys interpret this and what should we do to get those precious points in the exam?? 2010/6/28 Mouhammad Nasser engnasse...@hotmail.com Hi Kobel, The worst case takes a place upon the initialization of each RSVP call calculation, CUCM 7.0 LLD refers that amond N calls, it is recommended to calculate call number N as worst case, so it always succeeds (written in P.3-64 Configuration Recommendation) Regarding the number of bytes in FRF.12 header, do you recommend we always consider it a 4 Bytes? It is not mentioned in CUCM LLD, and I saw it fixed at 8 bytes in QoS LLD, I think it is better to go with 8 I don't know. I hope someone from IPExpert to explain this more, Amy: we shall be waiting for your kind reply here Finally, I know that both LLQ and ip rsvp bandwith values must be identical and calculated as = (N-1) calls at 20 mSec + 1 call at 10 mSec Thank you a lot in advance -- Hotmail: Trusted email with powerful SPAM protection. Sign up now.https://signup.live.com/signup.aspx?id=60969 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Presence Issues Softphone mode in CUPC Lab 5 Volume 2
Hello all Out of ideas now after troubleshooting extensively a Presence problem. I'm finalizing lab 5 Vol 2 and I can't get CUPC to download its TFTP configuration file from CUCM and for that reason I do not even see the option for selecting softphone control Any help is appreciated. What I have and what I've done is the following: 1- Cretaed device named UPC+12alphanumeric characters, in my case UPCTERRELLEPRYO, associated its line to the enduser 2- End user configured with primary extension, associated with UPC phone device, CTI control of its devices and group association to CTI enabled group. 3- Still in CUCM, Capabilities Assignment was provided for the user. 5- In CUPS, Application-Cisco Unified Personal Comm - Settings I have provided the IP addresses for TFTP server primary and secondary Presence status is working fine and Deskphone control works fine as well. My problem here is that the CUPC SIP phone is not getting in Show Server Health a tftp file to download. It displays the IP addres of TFTP primary and seoondary but it does not display the UPCTERRELLEPRYO.CNF.XML file to download. To troubleshoot this I have done the following: 1- Went in DOS and did a tftp -i 10.10.210.10 get UPCTERRELLEPRYO.CNF.XML and files downloaded OK so there is no network issues here. Inside the file I saw references to TFTP server as IP addresses so no name resolution issues either. 2- Ran Wireshark and did not see any attempts from the client machine to register with CUCM via SIP so client is not even attempting to register. In fact nothing displays when I filter the capture by the CUCM ip addresses. 3- Listing my cupc users by clicking in CUPS, application, Cisco Unified personal comm, user settings I see my users listed there but under the column Client Type nothing displays 4- Created another UPC device for another user with another name and it still presents same problem. 5- Tried to enable all phone tracing in CUCM and everything else related to SIP under trace settings and nothing displayed with relation to the UPC phone attempting to register. Not sure what to do next, my version of CUPC is 7.0(2.13496) I havem't looked for bugs yet. What version are you guys using? If anyone has any ideas please let me know ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Presence Issues Softphone mode in CUPC Lab 5 Volume 2
Thanks for your replies. Primary extension is assigned to end user and that extension matches with the line number of CUPC. The users are assigned to the Standard CCM End Users, and CTI Enabled groups What is the version of CUPC you guys use? Thank you On Sun, Jun 27, 2010 at 9:03 AM, kobel findko...@gmail.com wrote: See if adding the end user to Standard CUCM users group in CUCM helps regards On Sat, Jun 26, 2010 at 10:30 PM, Daniel Berlinski dberlin...@gmail.comwrote: Hello all Out of ideas now after troubleshooting extensively a Presence problem. I'm finalizing lab 5 Vol 2 and I can't get CUPC to download its TFTP configuration file from CUCM and for that reason I do not even see the option for selecting softphone control Any help is appreciated. What I have and what I've done is the following: 1- Cretaed device named UPC+12alphanumeric characters, in my case UPCTERRELLEPRYO, associated its line to the enduser 2- End user configured with primary extension, associated with UPC phone device, CTI control of its devices and group association to CTI enabled group. 3- Still in CUCM, Capabilities Assignment was provided for the user. 5- In CUPS, Application-Cisco Unified Personal Comm - Settings I have provided the IP addresses for TFTP server primary and secondary Presence status is working fine and Deskphone control works fine as well. My problem here is that the CUPC SIP phone is not getting in Show Server Health a tftp file to download. It displays the IP addres of TFTP primary and seoondary but it does not display the UPCTERRELLEPRYO.CNF.XML file to download. To troubleshoot this I have done the following: 1- Went in DOS and did a tftp -i 10.10.210.10 get UPCTERRELLEPRYO.CNF.XML and files downloaded OK so there is no network issues here. Inside the file I saw references to TFTP server as IP addresses so no name resolution issues either. 2- Ran Wireshark and did not see any attempts from the client machine to register with CUCM via SIP so client is not even attempting to register. In fact nothing displays when I filter the capture by the CUCM ip addresses. 3- Listing my cupc users by clicking in CUPS, application, Cisco Unified personal comm, user settings I see my users listed there but under the column Client Type nothing displays 4- Created another UPC device for another user with another name and it still presents same problem. 5- Tried to enable all phone tracing in CUCM and everything else related to SIP under trace settings and nothing displayed with relation to the UPC phone attempting to register. Not sure what to do next, my version of CUPC is 7.0(2.13496) I havem't looked for bugs yet. What version are you guys using? If anyone has any ideas please let me know ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Presence Issues Softphone mode in CUPC Lab 5 Volume 2
Hi Kobel Owner was setup for the mobility section to work. It is in there. Hi Roger The way I know how to verify dbReplication is: admin:show perf query class Number of Replicates Created and State of Replication ==query class : - Perf class (Number of Replicates Created and State of Replication) has instances and values: ReplicateCount - Number of Replicates Created = 412 ReplicateCount - Replicate_State= 2 My reading of this is that is all good. Am I right? Well, I have rebooted this many times already so I think I will just upgrade the client and see what happens. Will update you all. Thnaks 2010/6/27 Roger Källberg roger.kallb...@cygate.se Try to verify if db replication is ok, if not, fix that. You might also want to restart the CTI Manager on both sub and pub. Brgds, *Roger Källberg* CCIE #26199 (Voice) Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ Direkt: +46108787498 Växel: +46108787400 roger.kallb...@cygate.se -- *Från:* Daniel Berlinski [dberlin...@gmail.com] *Skickat:* den 26 juni 2010 23:18 *Till:* kobel *Kopia:* osl osl *Ämne:* Re: [OSL | CCIE_Voice] Presence Issues Softphone mode in CUPC Lab 5 Volume 2 Thanks for your replies. Primary extension is assigned to end user and that extension matches with the line number of CUPC. The users are assigned to the Standard CCM End Users, and CTI Enabled groups What is the version of CUPC you guys use? Thank you On Sun, Jun 27, 2010 at 9:03 AM, kobel findko...@gmail.com wrote: See if adding the end user to Standard CUCM users group in CUCM helps regards On Sat, Jun 26, 2010 at 10:30 PM, Daniel Berlinski dberlin...@gmail.com wrote: Hello all Out of ideas now after troubleshooting extensively a Presence problem. I'm finalizing lab 5 Vol 2 and I can't get CUPC to download its TFTP configuration file from CUCM and for that reason I do not even see the option for selecting softphone control Any help is appreciated. What I have and what I've done is the following: 1- Cretaed device named UPC+12alphanumeric characters, in my case UPCTERRELLEPRYO, associated its line to the enduser 2- End user configured with primary extension, associated with UPC phone device, CTI control of its devices and group association to CTI enabled group. 3- Still in CUCM, Capabilities Assignment was provided for the user. 5- In CUPS, Application-Cisco Unified Personal Comm - Settings I have provided the IP addresses for TFTP server primary and secondary Presence status is working fine and Deskphone control works fine as well. My problem here is that the CUPC SIP phone is not getting in Show Server Health a tftp file to download. It displays the IP addres of TFTP primary and seoondary but it does not display the UPCTERRELLEPRYO.CNF.XML file to download. To troubleshoot this I have done the following: 1- Went in DOS and did a tftp -i 10.10.210.10 get UPCTERRELLEPRYO.CNF.XML and files downloaded OK so there is no network issues here. Inside the file I saw references to TFTP server as IP addresses so no name resolution issues either. 2- Ran Wireshark and did not see any attempts from the client machine to register with CUCM via SIP so client is not even attempting to register. In fact nothing displays when I filter the capture by the CUCM ip addresses. 3- Listing my cupc users by clicking in CUPS, application, Cisco Unified personal comm, user settings I see my users listed there but under the column Client Type nothing displays 4- Created another UPC device for another user with another name and it still presents same problem. 5- Tried to enable all phone tracing in CUCM and everything else related to SIP under trace settings and nothing displayed with relation to the UPC phone attempting to register. Not sure what to do next, my version of CUPC is 7.0(2.13496) I havem't looked for bugs yet. What version are you guys using? If anyone has any ideas please let me know ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Presence Issues Softphone mode in CUPC Lab 5 Volume 2
Hello Pavan CUPC is not even requesting the config xml file, checked with wireshark. In show server health there is no value against TFTP.Filename= I can't get it to work even after the client upgrade. I guess I will re-image the CUPS server and will update later. Cheers 2010/6/27 Pavan pav.c...@gmail.com Daniel, Before you go check replication, check to see if cups is even requesting the correct config xml file. Replication could only be a problem when cups tries to register to ucm and ucm rejects the register request Sent from my phone On Jun 26, 2010, at 16:52, Roger Källberg roger.kallb...@cygate.se wrote: Hi Daniel, It's not always that you can trust the information given by the show perf query class Number of Replicates Created and State of Replication command. One easy thing that you can do to verify if you have a db repl problem is to put your phones, or any other device, in a pub only enviroment. If all works then you know that the sub didn't have the correct info. And in thet case you need to repair the db replication by utils debreplication stop ,1 on sub, then when promtpt returns on the sub put in the same command on pub). When the prompt returns on the pub use utils dbreplication repair all on the pub. This will take some time to complete. *Roger Källberg* CCIE #26199 (Voice) Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ Direkt: +46108787498 Växel: +46108787400 roger.kallb...@cygate.se -- *Från:* Daniel Berlinski [dberlin...@gmail.com] *Skickat:* den 26 juni 2010 23:44 *Till:* Roger Källberg *Kopia:* kobel; osl osl *Ämne:* Re: [OSL | CCIE_Voice] Presence Issues Softphone mode in CUPC Lab 5 Volume 2 Hi Kobel Owner was setup for the mobility section to work. It is in there. Hi Roger The way I know how to verify dbReplication is: admin:show perf query class Number of Replicates Created and State of Replication ==query class : - Perf class (Number of Replicates Created and State of Replication) has instances and values: ReplicateCount - Number of Replicates Created = 412 ReplicateCount - Replicate_State= 2 My reading of this is that is all good. Am I right? Well, I have rebooted this many times already so I think I will just upgrade the client and see what happens. Will update you all. Thnaks 2010/6/27 Roger Källberg roger.kallb...@cygate.se roger.kallb...@cygate.se Try to verify if db replication is ok, if not, fix that. You might also want to restart the CTI Manager on both sub and pub. Brgds, *Roger Källberg* CCIE #26199 (Voice) Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ Direkt: +46108787498 Växel: +46108787400 roger.kallb...@cygate.se -- *Från:* Daniel Berlinski [ dberlin...@gmail.comdberlin...@gmail.com] *Skickat:* den 26 juni 2010 23:18 *Till:* kobel *Kopia:* osl osl *Ämne:* Re: [OSL | CCIE_Voice] Presence Issues Softphone mode in CUPC Lab 5 Volume 2 Thanks for your replies. Primary extension is assigned to end user and that extension matches with the line number of CUPC. The users are assigned to the Standard CCM End Users, and CTI Enabled groups What is the version of CUPC you guys use? Thank you On Sun, Jun 27, 2010 at 9:03 AM, kobel findko...@gmail.com findko...@gmail.com wrote: See if adding the end user to Standard CUCM users group in CUCM helps regards On Sat, Jun 26, 2010 at 10:30 PM, Daniel Berlinski dberlin...@gmail.com dberlin...@gmail.com wrote: Hello all Out of ideas now after troubleshooting extensively a Presence problem. I'm finalizing lab 5 Vol 2 and I can't get CUPC to download its TFTP configuration file from CUCM and for that reason I do not even see the option for selecting softphone control Any help is appreciated. What I have and what I've done is the following: 1- Cretaed device named UPC+12alphanumeric characters, in my case UPCTERRELLEPRYO, associated its line to the enduser 2- End user configured with primary extension, associated with UPC phone device, CTI control of its devices and group association to CTI enabled group. 3- Still in CUCM, Capabilities Assignment was provided for the user. 5- In CUPS, Application-Cisco Unified Personal Comm - Settings I have provided the IP addresses for TFTP server primary and secondary Presence status is working fine and Deskphone control works fine as well. My problem here is that the CUPC SIP phone is not getting in Show Server Health a tftp file to download. It displays the IP addres of TFTP primary and seoondary but it does not display the UPCTERRELLEPRYO.CNF.XML file to download. To troubleshoot this I have done the following: 1- Went in DOS and did a tftp -i 10.10.210.10 get UPCTERRELLEPRYO.CNF.XML and files downloaded OK so there is no network issues here. Inside the file I saw references to TFTP server as IP addresses so no name resolution
Re: [OSL | CCIE_Voice] Vol2 Lab1, 4.2 Supplementary services
David please check the link below http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg17219.html These were the troubleshooting I completed with the help of other members of this list to get it working. I got it working by using IOS software MTP, adding some h323 commands under the CUBE, and unchecking wait for capabilities exchange from the h225 controlled trunk Cheers On Sun, Jun 27, 2010 at 12:18 PM, David Lee d16...@gmail.com wrote: Let me clarify. I am using IOS Software MTP, not the UCM software MTP. On Sat, Jun 26, 2010 at 8:04 PM, Pavan pav.c...@gmail.com wrote: You will need a hardware mtp (i.e dsp / ios sw ) not the one provided by ucm Sent from my phone On Jun 26, 2010, at 18:47, David Lee d16...@gmail.com wrote: Hello, I am at a lost. I got most of this section working. I can resume a call if the hold was initiated by an UCM phone or the CME SCCP phone. However, I cannot resume if the hold was initiated by the CME SIP phone. Any one have ideas what can be looked at to troubleshoot? (Software MTP is configured and active during the call. Codecs are also right.) Thanks, -Dave ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] HomeLab Equipments
Can you run 12.4(20T2) IOS code on them? DSP Farm configuration will be a bit different as well. On Sat, Jun 26, 2010 at 10:34 PM, Ken Tan thinkc...@gmail.com wrote: Hi, Can anyone advise if I can build a CCIE Voice homelab based on 3640 instead of 2811. I checked cisco web site it seems 3640 together with NM-HV PVDM-12 and VWIC-2MFT-E1/T1 seems workable. Had too many 3640 lying around do not wish to invest unnecessary. Any advise is greatly appreciated. Ken ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Presence Issues Softphone mode in CUPC Lab 5 Volume 2
Hi Guys I want to let you all know that it works only if the username is lower then 12 characters and match exactly with the UPCdevicename. Also for you info, phone Owner ID and assigning the end user to the CCM End users and CTI enabled groups was not neceessary to make it work. Cheers On Sun, Jun 27, 2010 at 12:16 PM, Daniel Berlinski dberlin...@gmail.comwrote: Hello Pavan CUPC is not even requesting the config xml file, checked with wireshark. In show server health there is no value against TFTP.Filename= I can't get it to work even after the client upgrade. I guess I will re-image the CUPS server and will update later. Cheers 2010/6/27 Pavan pav.c...@gmail.com Daniel, Before you go check replication, check to see if cups is even requesting the correct config xml file. Replication could only be a problem when cups tries to register to ucm and ucm rejects the register request Sent from my phone On Jun 26, 2010, at 16:52, Roger Källberg roger.kallb...@cygate.se wrote: Hi Daniel, It's not always that you can trust the information given by the show perf query class Number of Replicates Created and State of Replication command. One easy thing that you can do to verify if you have a db repl problem is to put your phones, or any other device, in a pub only enviroment. If all works then you know that the sub didn't have the correct info. And in thet case you need to repair the db replication by utils debreplication stop ,1 on sub, then when promtpt returns on the sub put in the same command on pub). When the prompt returns on the pub use utils dbreplication repair all on the pub. This will take some time to complete. *Roger Källberg* CCIE #26199 (Voice) Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ Direkt: +46108787498 Växel: +46108787400 roger.kallb...@cygate.se -- *Från:* Daniel Berlinski [dberlin...@gmail.com] *Skickat:* den 26 juni 2010 23:44 *Till:* Roger Källberg *Kopia:* kobel; osl osl *Ämne:* Re: [OSL | CCIE_Voice] Presence Issues Softphone mode in CUPC Lab 5 Volume 2 Hi Kobel Owner was setup for the mobility section to work. It is in there. Hi Roger The way I know how to verify dbReplication is: admin:show perf query class Number of Replicates Created and State of Replication ==query class : - Perf class (Number of Replicates Created and State of Replication) has instances and values: ReplicateCount - Number of Replicates Created = 412 ReplicateCount - Replicate_State= 2 My reading of this is that is all good. Am I right? Well, I have rebooted this many times already so I think I will just upgrade the client and see what happens. Will update you all. Thnaks 2010/6/27 Roger Källberg roger.kallb...@cygate.se roger.kallb...@cygate.se Try to verify if db replication is ok, if not, fix that. You might also want to restart the CTI Manager on both sub and pub. Brgds, *Roger Källberg* CCIE #26199 (Voice) Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ Direkt: +46108787498 Växel: +46108787400 roger.kallb...@cygate.se -- *Från:* Daniel Berlinski [ dberlin...@gmail.comdberlin...@gmail.com] *Skickat:* den 26 juni 2010 23:18 *Till:* kobel *Kopia:* osl osl *Ämne:* Re: [OSL | CCIE_Voice] Presence Issues Softphone mode in CUPC Lab 5 Volume 2 Thanks for your replies. Primary extension is assigned to end user and that extension matches with the line number of CUPC. The users are assigned to the Standard CCM End Users, and CTI Enabled groups What is the version of CUPC you guys use? Thank you On Sun, Jun 27, 2010 at 9:03 AM, kobel findko...@gmail.com findko...@gmail.com wrote: See if adding the end user to Standard CUCM users group in CUCM helps regards On Sat, Jun 26, 2010 at 10:30 PM, Daniel Berlinski dberlin...@gmail.com dberlin...@gmail.com wrote: Hello all Out of ideas now after troubleshooting extensively a Presence problem. I'm finalizing lab 5 Vol 2 and I can't get CUPC to download its TFTP configuration file from CUCM and for that reason I do not even see the option for selecting softphone control Any help is appreciated. What I have and what I've done is the following: 1- Cretaed device named UPC+12alphanumeric characters, in my case UPCTERRELLEPRYO, associated its line to the enduser 2- End user configured with primary extension, associated with UPC phone device, CTI control of its devices and group association to CTI enabled group. 3- Still in CUCM, Capabilities Assignment was provided for the user. 5- In CUPS, Application-Cisco Unified Personal Comm - Settings I have provided the IP addresses for TFTP server primary and secondary Presence status is working fine and Deskphone control works fine as well. My problem here is that the CUPC SIP phone is not getting in Show Server Health a tftp file to download. It displays the IP
Re: [OSL | CCIE_Voice] Connected number display
Manipulation at the route list level does not affect how the dialed number is updated on the phone display. I read this as per below: If I have two MGCP gateways BR1 and BR2, and call should go thru BR1 and if it fails it should go thru BR2. Requirement is if call goes through BR1, called number on my display should be 7 digits. If it goes thru BR2, called number should be 10 digits. How would manipulation at the route list help in this scenario? I have just tested here by manipulating the dialed number at the route pattern for the first choice gateway (MGCP BR1 - 7Digits) and by using called party xformation pattern for the second choice gateway (MGCP-BR2) In my case I could not do it for 10 digits because my BR2 router is in Spain. The phone display updates as per both transformation configs. If this is not correct please let me know what I'm missing Cheers On Tue, Jun 22, 2010 at 2:20 PM, Berry, Matthew J. mjbe...@krollontrack.com wrote: Daniel, You best bet would be to do the manipulation at the route list level for such a request. - Sent from my Blackberry -- *From*: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com *To*: Angel Perez gorr...@hotmail.com *Cc*: osl osl ccie_voice@onlinestudylist.com *Sent*: Mon Jun 21 16:04:44 2010 *Subject*: Re: [OSL | CCIE_Voice] Connected number display Hello Guys Just an idea and please ignore if this is a silly one or let me know if you have already tested this. Could you try to have your manipulation done at route pattern level for BR1 and for BR2 add a called party xformation in order to update the phone display when BR1 is down? As far as my understanding goes ANI manipulations at route pattern and (DNIS) called party transformation patterns applied to egress gateways will also have the cosmetic effect to phones screens. I will give this a go as soon as I have access to equipment again and will update Best Regards Daniel On Mon, Jun 21, 2010 at 11:13 PM, Angel Perez gorr...@hotmail.com wrote: Yes you are right, tested today, ccm engine will not try with another route pattern although controller/gw associated to the first rp is not up. I thought ccm would follow the same behaviour as a h323 gw. Since the only way I know to change phone display number is through route patt, my conclusion is that your requirements are not possible to be satified... Is this an exercise from a workbook or something you want to test? In case it's the first one let us know the solution becouse I can't think a way to make this work with ucm only. Thanks -- Date: Sun, 20 Jun 2010 17:28:59 +0530 Subject: Re: [OSL | CCIE_Voice] Connected number display From: voip.ccieci...@gmail.com To: gorr...@hotmail.com CC: siddas...@gmail.com; ccie_voice@onlinestudylist.com i tested bot the RP first.. then i did a no mgcp command on GW1 On Sun, Jun 20, 2010 at 4:52 PM, Angel Perez gorr...@hotmail.com wrote: Hi: Did you test both rp alone first to make sure it working correctly? Did you shutdown controller at br1 before testing backup path? thx -- Date: Sun, 20 Jun 2010 11:49:27 +0100 From: siddas...@gmail.com To: voip.ccieci...@gmail.com CC: gorr...@hotmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Connected number display Did you also try what I suggested? masking Called party at RL detail level! cisco voip wrote: I tried this just now. and it is not working, So what i was thinking is correct, it can match only one route pattern and call cannot come back. Is there any other way anyone would think of?? On Sun, Jun 20, 2010 at 12:00 AM, Angel Perez gorr...@hotmail.comwrote: Hi Ash, I think that to change calling number at phone display you may do transformation at rp level, correct me if i'm wrong thx -- Date: Sat, 19 Jun 2010 12:34:08 +0100 From: siddas...@gmail.com To: gorr...@hotmail.com CC: voip.ccieci...@gmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Connected number display Sorry Ignore my last post, I thought you are asking about Calling party number (ANI). The one Angel mentioned is a possible solution or try this one...make one route pattern, Create two RG in the RL, then place mask under Called party like XXX and XX under Route list detail level. I have not tested it so give it a try and let us know how it works. Ash Angel Perez wrote: Hi: The only way I can imagine to make this work is with to different route patterns, instead with one route pattern and a route list with two options, something like this: rp1: 91[2-9]XX.[2-9]XX DDI PREDOT, PT=br1-local-first-option rp2: 91.[2-9]XX[2-9]XX DDI PREDOT, PT=br1-local-sec-option br1 phone 1: css (phones,911,br1-local-first-option, br1-local-sec-option, ld, ...) Becouse rp1 and rp2 are
Re: [OSL | CCIE_Voice] Connected number display
Hello Guys Just an idea and please ignore if this is a silly one or let me know if you have already tested this. Could you try to have your manipulation done at route pattern level for BR1 and for BR2 add a called party xformation in order to update the phone display when BR1 is down? As far as my understanding goes ANI manipulations at route pattern and (DNIS) called party transformation patterns applied to egress gateways will also have the cosmetic effect to phones screens. I will give this a go as soon as I have access to equipment again and will update Best Regards Daniel On Mon, Jun 21, 2010 at 11:13 PM, Angel Perez gorr...@hotmail.com wrote: Yes you are right, tested today, ccm engine will not try with another route pattern although controller/gw associated to the first rp is not up. I thought ccm would follow the same behaviour as a h323 gw. Since the only way I know to change phone display number is through route patt, my conclusion is that your requirements are not possible to be satified... Is this an exercise from a workbook or something you want to test? In case it's the first one let us know the solution becouse I can't think a way to make this work with ucm only. Thanks -- Date: Sun, 20 Jun 2010 17:28:59 +0530 Subject: Re: [OSL | CCIE_Voice] Connected number display From: voip.ccieci...@gmail.com To: gorr...@hotmail.com CC: siddas...@gmail.com; ccie_voice@onlinestudylist.com i tested bot the RP first.. then i did a no mgcp command on GW1 On Sun, Jun 20, 2010 at 4:52 PM, Angel Perez gorr...@hotmail.com wrote: Hi: Did you test both rp alone first to make sure it working correctly? Did you shutdown controller at br1 before testing backup path? thx -- Date: Sun, 20 Jun 2010 11:49:27 +0100 From: siddas...@gmail.com To: voip.ccieci...@gmail.com CC: gorr...@hotmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Connected number display Did you also try what I suggested? masking Called party at RL detail level! cisco voip wrote: I tried this just now. and it is not working, So what i was thinking is correct, it can match only one route pattern and call cannot come back. Is there any other way anyone would think of?? On Sun, Jun 20, 2010 at 12:00 AM, Angel Perez gorr...@hotmail.com wrote: Hi Ash, I think that to change calling number at phone display you may do transformation at rp level, correct me if i'm wrong thx -- Date: Sat, 19 Jun 2010 12:34:08 +0100 From: siddas...@gmail.com To: gorr...@hotmail.com CC: voip.ccieci...@gmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Connected number display Sorry Ignore my last post, I thought you are asking about Calling party number (ANI). The one Angel mentioned is a possible solution or try this one...make one route pattern, Create two RG in the RL, then place mask under Called party like XXX and XX under Route list detail level. I have not tested it so give it a try and let us know how it works. Ash Angel Perez wrote: Hi: The only way I can imagine to make this work is with to different route patterns, instead with one route pattern and a route list with two options, something like this: rp1: 91[2-9]XX.[2-9]XX DDI PREDOT, PT=br1-local-first-option rp2: 91.[2-9]XX[2-9]XX DDI PREDOT, PT=br1-local-sec-option br1 phone 1: css (phones,911,br1-local-first-option, br1-local-sec-option, ld, ...) Becouse rp1 and rp2 are and equal match for UCM call processing engine, the pt orther will be the tie breaker, so the first choice would be rp1, and second choice would be rp2. Let us know how it goes Regards -- Date: Sat, 19 Jun 2010 16:01:09 +0530 From: voip.ccieci...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Connected number display Hi Experts, If I have two MGCP gateways BR1 and BR2, and call should go thru BR1 and if it fails it should go thru BR2. Requirement is if call goes through BR1, called number on my display should be 7 digits. If it goes thru BR2, called number should be 10 digits. From what i understand, display number is the manipulated number in Route Pattern. So I am not really sure how to change the display number on the basis of what gateway call is going out. Any Suggestions? -- Hotmail: Trusted email with powerful SPAM protection. Sign up now.https://signup.live.com/signup.aspx?id=60969 -- ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up now. https://signup.live.com/signup.aspx?id=60969 -- Hotmail:
Re: [OSL | CCIE_Voice] Gatekeeper CUBE - Pulling My Hair Out!
I believe you need a zone prefix for zone VIA. Have you tried to put that on? On Mon, Jun 21, 2010 at 12:21 PM, CCIE VOICE ccievoiced...@gmail.comwrote: Hey everyone...I have NO IDEA what is causing my issue and I was hoping for your assistance. I am currently working on Volume 2, Lab 1, Task 4.2 with no success. The goal is to dial 3XXX from HQ or BR1 and route the call from CUCM--GK--CUBE--BR2-RTR. I am getting the *Viazone gateway selection failed for zone VIA* error message. I have included the relevant configuration below. Any help is appreciated! ip domain name proctorlabs.com voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip fax protocol cisco interface Loopback0 ip address 10.10.110.1 255.255.255.255 h323-gateway voip interface h323-gateway voip id VIA ipaddr 10.10.110.1 1719 h323-gateway voip h323-id HQ-RTR sccp local FastEthernet0/0.20 sccp ccm 10.10.200.3 identifier 1 priority 1 version 7.0 sccp sccp ccm group 1 bind interface FastEthernet0/0.20 associate ccm 1 priority 1 associate profile 1 register HQ-XCODER dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 4 associate application SCCP dial-peer voice 3000 voip description GET CALL FROM GATEKEEPER incoming called-number 3... dial-peer voice 3001 voip description SEND CALL BACK TO GATEKEEPER destination-pattern 3...$ session target ras codec g711ulaw gatekeeper zone local UCM proctorlabs.com zone local VIA proctorlabs.com zone local UCME proctorlabs.com outvia VIA zone prefix UCM 1... zone prefix UCME 3... zone prefix UCM 5... gw-type-prefix 1#* default-technology no shutdown telephony-service sdspfarm units 5 sdspfarm transcode sessions 4 sdspfarm tag 1 HQ-XCODER max-ephones 1 max-dn 1 ip source-address 10.10.200.3 port 2000 max-conferences 8 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Jun 20 2010 23:20:02 ! HQ-RTR#sh gatek end GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags --- - --- - - - 10.10.110.2 1720 10.10.110.2 65228 VIA H323-GW H323-ID: BR1-RTR Voice Capacity Max.= Avail.= Current.= 0 10.10.110.3 1720 10.10.110.3 57209 UCME H323-GW H323-ID: BR2-RTR Voice Capacity Max.= Avail.= Current.= 0 10.10.200.3 1720 10.10.200.3 51074 VIA H323-GW H323-ID: HQ-RTR Voice Capacity Max.= Avail.= Current.= 0 192.168.1.1036728 192.168.1.1033279 UCM VOIP-GW H323-ID: gk-trunk_1 Voice Capacity Max.= Avail.= Current.= 0 192.168.1.1135438 192.168.1.1132790 UCM VOIP-GW H323-ID: gk-trunk_2 Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 5 HQ-RTR#debug gatek main 10 HQ-RTR# Jun 21 00:15:32.991: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Jun 21 00:15:32.995: ////GK/gk_rassrv_arq: arqp=0x47D96338,crv=0xB0, answerCall=0 Jun 21 00:15:32.995: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC Jun 21 00:15:32.995: //8000830CB000/8000830CB000/GK/gk_dns_query: No Name servers Jun 21 00:15:32.995: //8000830CB000/8000830CB000/GK/rassrv_get_addrinfo: (1#3001) Matched tech-prefix 1# Jun 21 00:15:32.995: //8000830CB000/8000830CB000/GK/rassrv_get_addrinfo: (1#3001) Matched zone prefix 3 and remainder 001 Jun 21 00:15:32.995: ////GK/gk_rassrv_get_ingress_network: returning default ingress network = 1 Jun 21 00:15:32.995: //8000830CB000/8000830CB000/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x4A68299C Jun 21 00:15:32.995: //8000830CB000/8000830CB000/GK/rassrv_arq_select_viazone: matched zone is UCM, and z_invianamelen=0 Jun 21 0 HQ-RTR#0:15:32.995: //8000830CB000/8000830CB000/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x4B9C9910 Jun 21 00:15:32.995: //8000830CB000/8000830CB000/GK/rassrv_arq_select_viazone: matched zone is UCME, and z_outvianamelen=3 Jun 21 00:15:32.995: //8000830CB000/8000830CB000/GK/rassrv_arq_select_viazone and z_outvianamep=VIA Jun 21 00:15:32.995: //8000830CB000/8000830CB000/GK/rassrv_arq_select_viazone: Received ARQ for a zone (UCME) that has an outviazone (VIA) specified. Pick an IP-IP gateway in that viazone. Jun 21 00:15:32.995: ////GK/gk_gw_select_ipipgw_random: zonep: 0x4A682E7C, tpp: 0x4A62E180, current_endpt: 0 Jun 21 00:15:32.995: ////GK/gk_gw_select_ipipgw_random: Gateway
[OSL | CCIE_Voice] Lab 3 Volume 2 CUE-CUCM JTAPI integration and MWI
Hello Could someone clarify the mechanism behind MWI on/off in this CTI integration scenario? It works even without assigning the end device (phone of CUE subscriber) to the cti application user. I’m reviewing the steps I took to complete the lab and can’t see what was done to have it working. Thank you ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Lab 3 Volume 2 SRST CUE Unknown caller
Hello In the following scenario: Phone 1002 rings 3002 in SRST mode, calls are unanswered and forwarded to CUE. I leave a msg for 3002 and when collecting it the following is played by CUE “from unknown caller”. I see the call is sent to CUE as follows: From: BR1PH2 sip:+16178631...@10.10.202.1sip%3a%2b16178631...@10.10.202.1 To: sip:3...@10.10.202.2 sip%3a3...@10.10.202.2 I would like to configure it so that CUE plays “from 1002” instead. What configuration is required to achieve this? Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME CUCM call hold problems
Hello Angel Thanks a lot for this it has worked by configuring IOS MTP. May I ask you if call transfers worked fine for you as well? In my setup call transfers are not working properly. If for instance I send a call from a CME phone to a CUCM phone then press transfer, the CME phone remains on hold after call is completed with the transfer-to party. The only way to complete transfer is by pressing hold twice on the CME phone. Anyone got call transfers to work perfectly? Same behaviour seen with Supervised or Blind xfer. My CME configs as follows: voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip supplementary-service h450.12 This was added in an attempt to get call xfers to work flawlessly h323 emptycapability This was added in an attempt to get call xfers to work flawlessly h225 id-passthru This was added in an attempt to get call xfers to work flawlessly h225 connect-passthru This was added in an attempt to get call xfers to work flawlessly no call service stop h245 passthru tcsnonstd-passthru This was added in an attempt to get call xfers to work flawlessly sip bind control source-interface Vlan400 bind media source-interface Vlan400 registrar server ! telephony-service sdspfarm units 1 sdspfarm transcode sessions 3 sdspfarm tag 1 br2-xcoder no auto-reg-ephone load 7960-7940 P00308000500 load 7965 SCCP45.8-3-3S max-ephones 3 max-dn 6 no-reg ip source-address 10.10.110.3 port 2000 time-format 24 date-format dd-mm-yy max-conferences 8 gain -6 call-forward pattern .T transfer-system full-consult transfer-pattern .T create cnf-files version-stamp 7960 Jun 11 2010 08:11:35 ! sccp local Vlan400 sccp ccm 10.10.110.3 identifier 1 version 5.0.1 sccp ip precedence 3 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register br2-xcoder signaling dscp af31 ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 maximum sessions 19 associate application SCCP Cheers On Thu, Jun 10, 2010 at 8:12 PM, Angel Perez gorr...@hotmail.com wrote: Hi: You need software mtp from ios not from ucm, make sure that ios mtp are configured and registered, to be sure that mtp is working verify it with sh sccp or from ucm. Once you have ios mtp registered add a mrg and include all ucm software mtp and cnf, then *do not* include this mrg to any mrgl, this way you will be sure that this resources are not available for your trunk/phones. Also be sure that in the trunk/phones mrgl the ios mtp rosource is above other ucm software resources. Then place a call, press hold and verify with sh sccp con For more information check: CUCM 7 SRND page 5-11 (H.323 Trunks with Media Termination Points ) http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg15970.html hth -- Date: Thu, 10 Jun 2010 19:17:48 +1200 From: dberlin...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME CUCM call hold problems Hello all After completing lab 2 of volume 2 Gatekeeper section I found the following behaviour when testing call hold between phones registered to CUCM and CME respectively: By saying successful I mean the ability to place call on hold and resume Calls from CUCM phones bound for CME phones placed on hold by either CUCM or CME phones are successful Calls from CME phones bound for CUCM phones placed on hold by CUCM phones are not successful. The problem manifestates as not allowing me to resume the call. Same scenario but pushing hold from a CME phone is successful. With this scenario in mind the following was done: MTP required checkbox in trunk is checked and added to MRL of trunk's device pool and the trunk page itself, software MTPs and Hardware IOS xcoders While testing with these Media Resources configured show perf query class counters were not incrementing at all when I pushed hold on the CUCM phone - I was expecting to see MTP usage once pushing the hold.button - Am I right to expect it to happen? show sccp connections did not show anything either as I thought that the xcoder was being used instead. In addition, wait for TCS on trunk were unchecked and outbound faststart was also configured as last resort to see if any difference could be seen in behaviour. rebooting servers did not help either. Anyone experienced this? Cheers -- Hotmail: Powerful Free email with security by Microsoft. Get it now.https://signup.live.com/signup.aspx?id=60969 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CME CUCM call hold problems
Hello all After completing lab 2 of volume 2 Gatekeeper section I found the following behaviour when testing call hold between phones registered to CUCM and CME respectively: By saying successful I mean the ability to place call on hold and resume Calls from CUCM phones bound for CME phones placed on hold by either CUCM or CME phones are successful Calls from CME phones bound for CUCM phones placed on hold by CUCM phones are not successful. The problem manifestates as not allowing me to resume the call. Same scenario but pushing hold from a CME phone is successful. With this scenario in mind the following was done: MTP required checkbox in trunk is checked and added to MRL of trunk's device pool and the trunk page itself, software MTPs and Hardware IOS xcoders While testing with these Media Resources configured show perf query class counters were not incrementing at all when I pushed hold on the CUCM phone - I was expecting to see MTP usage once pushing the hold.button - Am I right to expect it to happen? show sccp connections did not show anything either as I thought that the xcoder was being used instead. In addition, wait for TCS on trunk were unchecked and outbound faststart was also configured as last resort to see if any difference could be seen in behaviour. rebooting servers did not help either. Anyone experienced this? Cheers ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Lab 1 Volume 2 questions
Hello all Hope you all doing well. I would like to bring to you guys attention and hopefully get some interesting replies on some technology topics of Lab 1 of Volume 2 that I am not 100% sure about. 1- Gatekeeper section: Had a problem with the calls between CME and CUCM taking all WAN bandwidth overtime. This was solved after completing the CAC section by issuing “bandwidth zone UCM 32” command in gatekeeper. That being said a couple of things come to mind: Show gatekeeper calls does not show the same output as asked back in sections 4.2 and 4.3 of the lab, secondly PG suggests that the CAC sestion could also be solved by issuing a gatekeeper command for the CME zone but that would be 240Kbps. I did not understand why this was suggested as I believe we have 2 call legs here. Right? For reference this was mentioned on page 99 of the Proctor Guide. I’m particularly interested in clarifying this question because I suspect I’m missing something fundamental here. My understanding is that we are talking 2 g711ulaw call legs over the WAN between the CUBE and CME right? 2- It is not clear why it is suggested not to include g729r8 in IOS xcoder configuration as I believe this is necessary in situations where g729r8 is the codec that needs xcoding 3- Attendant Console question what is the expected behaviour while testing? I ring the pilot point and it rings only in one of the 2 extensions never hunting over to the next. How did this work for you? Have you created users and logged in to the Attendant console CTI app to get it to hunt properly? 4- What keywords in the call routing/Device Mobility section defined the requirements for configuring the US sites in the same DMG? I decided to configure those 2 device pools in different DMGs because of the question stating neet not to keep class of restriction - I based my decision in configuring not to inherit roaming sensitve settings on that statement. 7 dgt ANI presentation without name for 911 calls was preserved becaue HQ LRG was used by BR1 Phone while roaming. Was it wrong? Best regards Daniel ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Codec selection question
Hello Ryan Based on what I think I learned correctly from the book Cisco IP Communications Express - CallManager Express with Cisco Unity Express this call will always be hairpinned by CME. CUCM uses the empty capability set standard (ECS) for its supplementary services while CME uses between few options, the H.450.3 standards for Call Forwards. This gets enabled by the telephony-services command call-forward pattern .T This is a method where you invoke the call replacement I believe is what you aiming for (where a signal is sent back to the calling party to re-originate the call to the forwarded-to destination) These two methods are not compatible. So as soon as your CME phone invokes the call transfer the following events happen: CME detects a call involving call manager by using special h323 IEs. If H450.3 is enabled then CME disables this mechanism and hairpins the call by using its built-in MTP resouces. So as far as I know your scenario cannot be achieved because as soon as the call is hairpinned then you have mismatching codecs configured between your CME dial-peers and the xcoder will be invoked. Please if someone disagrees speak up as I'm keen to confirm my understanding as well. Cheers Daniel On Tue, Jun 8, 2010 at 7:24 AM, Ryan Schwab schwab...@shaw.ca wrote: I have a real world example question that I'm hoping to gain a better insight/clarification to the Codec selection process on h323 dialpeers and CUCM regions. Scenario: An h323 trunk between a CUCM cluster and a CME/CUE ISR. Typical scenario of utilizing g729 over the WAN, but of course CUE can only use g711. Currently, a transcoder on the CME is allowing this to work properly. Question: Is there any way to configure this in such a way that when two IP phones connect, g729 is established, but if the CME phone forwards the call to the CUE Pilot, the CUCM Phone will establish g711 to the CUE eliminating the need for a transcoder? Comments/Questions are appreciated! Thanks, Ryan ___ 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] Volume 2 Lab 1 4.2 4.3
Hi Matthew I'm using my own setup. On Sun, Jun 6, 2010 at 12:32 PM, Matthew Berry ciscovoiceg...@gmail.comwrote: Are you using Proctor or your own lab setup? *Matthew Berry* *A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written* *Vitals:* *GVoice: *+1.612.424.5044 *Gmail*: ciscovoiceg...@gmail.com *Skype*: ciscovoiceguru *Twitter*: ciscovoiceguru *Cert Stats:* Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 6/5/2010 6:37 PM, Daniel Zeiger Berlinski wrote: Hello there I have completed the gatekeeper routing section of this lab and while testing I noticed that everytime I ring any BR2 phones from either HQ or BR1 using g711ulaw from CUBE to CME the call drops after 1 minute apprx. Looking further I noticed that all WAN bandwidth I have, is taken to the point that OSPF adjacency is lost. (in the case of my devices I have 128Kbps for these Frame tails because of hardware limitations of my lab) Well, show gatekeeper call displays exactly how the question mandates and supplementary services such as hold work as well but just for apprx 1 minute for the reasons I mentioned before. If I hop on my Frame switch I see the bandwidth consumption going higher and higher as time elapses. I'm running this setup with 2801 routers and 12.4(20)T2 advanced enterprise code. In essence what I'm seeing here is g711/g729 calls are consuming bandwidth until no more WAN bandwidth is available. I am starting to suspect of this being bug related? I'm not able to see the reason behind such behaviour and would be greatful if someone could help. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUBE hold-resume issue
Hello Angel Can you tell us whether or not you have MTP checked in your gK controlled trunk and a MTP resource available for it to invoke. Thanks On Wed, Apr 14, 2010 at 2:33 AM, Angel Perez gorr...@hotmail.com wrote: Hi all: I've the following scenario: sip phone(3001)---cme---cube---ccm---sip phone(1001) These three gws (cme, cube, ccm) are registered to hq gk Calls are working as expected, (codec, etc) the only thing that doesn't work is that when I've a call between 1001 and 3001 I can hold the call from 1001 but the I can't resume it (if i try to resume the calls stays on hold and then it disconnect), if i hold the call from 3001 I can resume with no problems Any suggestion? -- Hotmail: Powerful Free email with security by Microsoft. Get it now.https://signup.live.com/signup.aspx?id=60969 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
