Re: [OSL | CCIE_Voice] CBarge in SRST ~(Again)
My configuration has worked. You need to make sure that the ephone configuration has privacy off in order for the cBarge to work with auto provision none. Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.commailto:david.ra...@kroll.com From: Bryan [mailto:ccieiwi...@gmail.com] Sent: Wednesday, July 21, 2010 8:28 AM To: Berry, Matthew J. Cc: Mark Holloway; osl osl Subject: Re: [OSL | CCIE_Voice] CBarge in SRST ~(Again) Sorry to jump in on the topic. Matt, just curious were you successful with this configuration? It does not work for me with auto-provision none and an ephone-template under the srst ephone template configuration. Another strange thing I have noticed in SRST is when I issue a show telephony-service all, and scroll down to the ephone-template section. It says privacy default, and I have not figured out how to get rid of it or if it is even possible. On Wed, Jul 21, 2010 at 9:20 AM, Berry, Matthew J. mjbe...@krollontrack.commailto:mjbe...@krollontrack.com wrote: Mark - Try removing all your learned ephone configuration, change the auto provision mode to none, and then add the ephone template under srst ephone template. See if that will work for you. Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.commailto:david.ra...@kroll.com From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Holloway Sent: Wednesday, July 21, 2010 12:33 AM To: Mark Holloway Cc: osl osl Subject: Re: [OSL | CCIE_Voice] CBarge in SRST ~(Again) I found this blog post showing an example config that I have loaded on my router that I think should be the correct way to configure cbarge in srst. However, I still can't get it to work. When my phones 'fallback' and I call from the pstn into the shared line, the other phone can't barge the call because when I go off-hook on the shared line the 'cBarge' softkey will display for a fraction of a second the it turns into a ghosted 'Redial' softkey. http://ccieash.wordpress.com/2010/06/21/hardware-conferencing-ios/ On Jul 20, 2010, at 2:30 PM, Mark Holloway wrote: The odd part is this.. Once the phones fall back to srst and I place a call from the PSTN to 2005, I go to the second phone and press the second line key for 2005. I expect the phone to display Remote in Use and offer the cBarge and NewCall softkeys. However, I can see the cBarge and NewCall softkeys appear for a split second, then they disappear and the normal softkeys (CallFwd, etc) appear. On Jul 20, 2010, at 2:18 PM, Mark Holloway wrote: Angel - What kind of phones did you test srst cBarge with? I can't get this to work with my 7965 phones. I needed to add more details under the ephone configuration in order for ephone-template 1 to be applied to the phone which should make the cBarge softkey available during srst. Otherwise, if I reference telephony-service 'srst ephone template 1' it doesn't seem to load properly on the 7965's when they fall back. Only by explicitly assigning the ephone-template 1 under the ephone works (which requires the mac address to be assigned as well). Even so, when a call comes in from the PSTN to my shared line 2005 during srst, I cannot get the other phone to display the cBarge softkey. When I go off-hook on the second phone on line 2005, I get dial-tone but the phone is treating this like any normal DN wanting to make an outbound call. I have made sure Privacy = off but still no luck. telephony-service sdspfarm units 1 sdspfarm tag 1 BR1-CONF conference hardware srst mode auto-provision none srst dn line-mode dual-octo max-ephones 2 max-dn 20 no-reg primary ip source-address 192.168.1.254 port 2000 system message SRST MODE time-zone 8 voicemail 917752011015 max-conferences 8 gain -6 call-forward pattern .T transfer-system full-consult transfer-pattern .T r2-br1(config)#do sh sccp SCCP Admin State: UP Gateway Local Interface: Vlan10 IPv4 Address: 192.168.1.254 Port Number: 2000 IP Precedence: 5 User Masked Codec list: None Call Manager: 192.168.1.254, Port Number: 2000 Priority: 1, Version: 7.0, Identifier: 1 Conferencing Oper State: ACTIVE - Cause Code: NONE Active Call Manager: 192.168.1.254, Port Number: 2000 TCP Link Status: CONNECTED, Profile Identifier: 1 ephone-template 1 privacy off privacy-button conference drop-mode local softkeys remote-in-use CBarge Newcall softkeys idle Redial Newcall Cfwdall softkeys seized Cfwdall Endcall Meetme Pickup Redial softkeys connected Hold Endcall Trnsfer Park Confrn ConfList Select Join ephone-dn 1 octo-line number 2001 no-reg primary label 2001 description 7753012001 name Br1Ph1 call-forward busy 917752011015 call-forward noan 917752011015 timeout 20 ephone-dn 2 octo-line number 2002 no-reg primary label 2002
Re: [OSL | CCIE_Voice] Voice L_A_B Q_U_E_R_I_E_S
Is someone going to ban this guys' email for sharing NDA material on this list? I don't care what people choose to share offline, that's their business. But don't blast this stuff out for the whole world to see. Thanks for posting it (sarcasm). I bet these questions will be removed from the lab exam now. Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.com -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccieid1ot Sent: Wednesday, July 21, 2010 11:01 AM To: Randall Saborio Cc: ccie_voice@onlinestudylist.com; voicerack voicerack Subject: Re: [OSL | CCIE_Voice] Voice L_A_B Q_U_E_R_I_E_S Wow, I just reread the question. Looks like we need to know the CUCM cli to achieve this. H, I have to try this out. On Wed, Jul 21, 2010 at 10:55 AM, Randall Saborio ill2...@gmail.com wrote: What do you mean by using SSH for changing the background images. SSH to where? To the IP Phone? I know you can ssh to the phone and do some crazy debugging, but have no clue what you are talking about here. On Wed, Jul 21, 2010 at 9:05 AM, voicerack voicerack voicer...@gmail.com wrote: Hi, QUESTION 1) Background image Background image on CME PHONES Cisco do not want these files to be uploaded from CUCM but via SSH :- HOW TO ACHIEVE THIS WITHOUT UPLOADING TO CUCM TFTP MY COMMENTS Why the hell cisco want us to use 3rd party when we can use CUCM?? Why they have nt mentioned the use of SSH in the blue-pint?? How to achieve this without using cucm TFTP?? 2) MEET ME LA-PH1 only can initiate the meet me conference The other users can call the meet me number and get connected to the conference. PSTN can also access the conference bridge/ 1234 is the number for the meet me. Make sure when user join and leave the conference beeps are heard SOLUTION 1 ephone-dn 7 octo-line number 1234 no-reg both conference meetme no huntstop ephone-dn 8 octo-line number 1234 no-reg both conference meetme preference 1 voice class custom-cptone leavetone dualtone conference frequency 400 800 cadence 400 50 200 50 ! voice class custom-cptone jointone dualtone conference frequency 600 900 cadence 300 150 dspfarm profile 2 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 3 conference-join custom-cptone jointone conference-leave custom-cptone leavetone associate application SCCP SOLUTION 2 ephone-dn 7 octo-line number 1234 no-reg both conference meetme voice class custom-cptone leavetone dualtone conference frequency 400 800 cadence 400 50 200 50 200 50 ! voice class custom-cptone jointone dualtone conference frequency 600 900 cadence 300 150 300 100 300 50 dspfarm profile 2 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 3 conference-join custom-cptone jointone conference-leave custom-cptone leavetone associate application SCCP FOR BOTH SOLUTION DISDO IS GIVING 0 why why why? 3) Presence a) LA-PH-2 should be able to monitor LA-PH1 line 1 their should be a 3rd line on LA-PH-2 that monitors this phone. When you push this button it should speed dial to 4001. when 4001 is on the phone this button should show red b. When LA-PH-1 line 1(4001) is on the phone you should see the status of this call in the local directory of phone 1 SOLUTION 1 Presence ip http server sip-ua presence enable presence presence call-list telephony-service directory entery 1 4001 name SC Phone 1 directory entery 2 4002 name SC Phone 2 url directories http://142.102.66.254/localdirectory ephone-dn 1 octo-line name SC Phone 1 allow watch ephone-dn 2 name SC Phone 2 allow watch ephone 2 blf-speed-dial 1 4001 label SpeedDial-4001 .SOLUTION 2 presence presence call-list ephone-dn 1 octo-line name LA Phone 1 allow watch ephone-dn 2 name LA Phone 2 allow watch ephone 2 Presenc call-list butt 1:2 2:4 3m1 AGAIN THE QUESTION WHY THE HELL DISDO IS GIVING THIS WRONG, AFTER ASKING TROCTOR he said if it is not define you can use any key word e.g 3w1 or blf speed dial Then why the hell we are not getting marks on the same. 4) QUESTION a) Queue 1 5 in the priority queue . queue 2 4,6,7 queue 3 2, 3 queue 4 0 b. guarantee Queue 1 has the 25% of the bandwidth. the other queues should share the bandwidth as 30 40 30. c. Once queue 2 reaches 60% capacity COS 4 packets should be dropped. SOLUTION mls qos mls qos map cos-dscp 0 8 16 24 32 46 48 56 mls qos srr-queue output cos-map queue 1 threshold 3 5 mls qos srr-queue output cos-map queue 2 threshold 1 4 mls qos srr-queue output cos-map queue 2 threshold 3 6 7 mls qos srr-queue output cos-map queue 3 threshold 3 2 3 mls qos srr-queue
Re: [OSL | CCIE_Voice] Call to PSTN fails - Volume 1 Lab 5C
Scott - You need to change the pri-group timeslots 1-3,23-24 to pri-group timeslots 1-3 (H.323 gateway) or pri-group timeslots 1-3 service mgcp (MGCP gateway). If you look at your running config, IOS will add the ,24 for the D channel of the circuit. However, if you try to copy your config and paste it into a blank router, the system will not see the 24 as the D channel. Instead, it will assume that you are using 1,2,3, and 24 as B CHANNELS. You also need a dial-peer for inbound POTS calls. Dial-peer voice 1 pots Direct-inward-dial Incoming called-number . Without this, you will be forced to use dial-peer zero which is a big no-no. Give it a shot and let me know. HTH Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.commailto:david.ra...@kroll.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Scott Newberry Sent: Thursday, July 08, 2010 11:01 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Call to PSTN fails - Volume 1 Lab 5C Hopefully someone can tell me that I'm off my rocker... I've spent several hours over the course of two nights looking at this, thinking that I MUST be doing something wrong, but the only answer I can come up with is that the PSTN config is off. This is using ProctorLabs.com, and IPx Vol 1 Lab 5C. I see the line about the fan that's failed, but was on a different pod of equipment last night, so I don't believe this to be hardware-related. Calls from HQ phones to 911 are failing. My PSTN phone rings for a partial ring, then my HQ-RTR sends a disconnect to the PSTN, with the message Cause i = 0x80AC - Requested circuit/channel not available. You'll notice that the PSTN is set up with the B-Channels split... pri-group timeslots 1-3,23-24... Is that a valid config? I've never seen that before, but assumed that I must be the problem. Initially I had my pri-group set up for 1-3,24, but then also changed it to match the PSTN config. I've tried ascending and descending for B-channel selection. Initially I was sending the digits 911 as unknown/unknown, but just for grins, tried translating that to ISDN/subscriber. Nothing changes the cause code that I send to the PSTN. Relevant portions of the config are below. Please, someone, tell me I'm an idiot, or tell me that the PSTN config is wrong. I'm leaning towards the latter (though the two options may not be mutually exclusive), but I just can't believe that I'm the first one to run up against a bad config on an early-stage lab, such that it wouldn't be changed already. Maybe you guys are all just jumping straight to Volume 2? Thanks! Scott PSTN Config ! voice translation-rule 212 rule 1 /^1212/ /1234/ rule 2 /^1/ /\0/ rule 3 /^394/ /\0/ rule 4 /^011/ /\0/ rule 5 /^212/ /\0/ rule 6 /^911$/ /\0/ rule 9 /./ /1234/ ! voice translation-profile block-call-into-HQ translate called 212 ! controller T1 0/3/0 framing esf clock source internal linecode b8zs pri-group timeslots 1-3,23-24 description ** T1 VOICE CONNECTION TO HQ-RTR ** ! interface Serial0/3/0:23 description ** T1 PRI D-CHANNEL TO HQ-RTR ** no ip address encapsulation hdlc isdn switch-type primary-ni isdn protocol-emulate network isdn incoming-voice voice isdn outgoing display-ie no cdp enable ! voice-port 0/3/0:23 translation-profile incoming block-call-into-HQ translation-profile outgoing display-proper-ani-into-HQ description ** PRI VOICE TRUNK TO HQ-RTR ** ! HQ Config ! voice translation-rule 911 rule 1 /911/ /911/ type any subscriber plan any isdn ! voice translation-profile 911_OUT translate called 911 ! controller T1 0/0/0 framing esf linecode b8zs pri-group timeslots 1-3,23-24 ! interface Serial0/0/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn bchan-number-order ascending isdn outgoing display-ie no cdp enable ! dial-peer voice 911 pots translation-profile outgoing 911_OUT destination-pattern 911 clid strip name port 0/0/0:23 forward-digits 3 PSTN - debug isdn q931 Jul 9 07:42:08.670: %ENVMON-3-FAN_FAILED: Fan 1 not rotating Jul 9 07:42:38.562: ISDN Se0/3/0:23 Q931: RX - SETUP pd = 8 callref = 0x008C Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Calling Party Number i = 0x2181, '2123945002' Plan:ISDN, Type:National Called Party Number i = 0xC1, '911' Plan:ISDN, Type:Subscriber(local) Jul 9 07:42:38.590: ISDN Se0/3/0:23 Q931: TX - CALL_PROC pd = 8 callref = 0x808C Channel ID i = 0xA98381 Exclusive, Channel 1 Jul 9 07:42:38.598: ISDN Se0/3/0:23 Q931: TX - ALERTING pd = 8 callref = 0x808C Progress Ind i = 0x8188 - In-band info or appropriate now available Jul 9 07:42:38.670: %ENVMON-3-FAN_FAILED: Fan 1 not rotating
Re: [OSL | CCIE_Voice] isdn plan
I set it for everything, but that's just me. Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.com -Original Message- From: Mark Holloway [mailto:m...@markholloway.com] Sent: Thursday, July 08, 2010 11:42 PM To: Berry, Matthew J. Cc: OSL osl Subject: Re: [OSL | CCIE_Voice] isdn plan Are you setting plan/type for both the called and calling numbers or just one of them? For example, if a task says the pstn provider wants the called party number type set and you set the plan/type for the called number, are you just leaving the calling portion set to CallManager or are you setting the plan/type for that as well? On Jul 7, 2010, at 11:43 AM, Berry, Matthew J. wrote: I make a habit of always setting the plan to ISDN. Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.com -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Holloway Sent: Wednesday, July 07, 2010 1:40 PM To: OSL osl Subject: [OSL | CCIE_Voice] isdn plan When tasked with setting the call type to unknown, subscriber, national, or international, are you guys also setting the plan to isdn or are you just specifying the type and leaving the plan as unknown even though all the pstn access is isdn? ___ 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] isdn plan
I make a habit of always setting the plan to ISDN. Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.com -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Holloway Sent: Wednesday, July 07, 2010 1:40 PM To: OSL osl Subject: [OSL | CCIE_Voice] isdn plan When tasked with setting the call type to unknown, subscriber, national, or international, are you guys also setting the plan to isdn or are you just specifying the type and leaving the plan as unknown even though all the pstn access is isdn? ___ 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] HQ BR2 - CUE Transcoding
Mark - Make sure that g729r8 is added under the dspfarm profile. Also, make sure you CUE dial-peer is hardcoded to be G711ulaw. Otherwise, it will try to use the default which is g729. Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.com -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Holloway Sent: Tuesday, July 06, 2010 12:44 PM To: Graham Hopkins Cc: OSL osl Subject: Re: [OSL | CCIE_Voice] HQ BR2 - CUE Transcoding Thanks, everyone. I configured the Transcoder locally on BR2. Now my issue is when I call from HQ to BR2 CUE, the call is answered by CUE but I do not hear the CUE attendant. The HQ phone shows RTP Sender packets incrementing but my Rcvr packets is not incrementing. Local BR2 phones work fine, so I know CUE is up and running. Has anyone experienced one-way audio with CUE before while Transcoding? r3-br2#show sccp Transcoding Oper State: ACTIVE - Cause Code: NONE Active Call Manager: 192.168.1.254, Port Number: 2000 TCP Link Status: CONNECTED, Profile Identifier: 2 Reported Max Streams: 8, Reported Max OOS Streams: 0 Supported Codec: g711ulaw, Maximum Packetization Period: 30 Supported Codec: g711alaw, Maximum Packetization Period: 30 Supported Codec: g729ar8, Maximum Packetization Period: 60 Supported Codec: g729abr8, Maximum Packetization Period: 60 Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30 Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30 Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30 r3-br2#show sdspfarm units mtp-3 Device:CME-XCODE TCP socket:[7] REGISTERED in SCCP ver 17/10 actual_stream:8 max_stream 8 IP:192.168.1.254 31790 MTP Dixieland keepalive 19 Supported codec: G711Ulaw G711Alaw G729a G729ab r3-br2# show run | sec teleph telephony-service sdspfarm units 5 sdspfarm transcode sessions 6 sdspfarm tag 2 CME-XCODE r3-br2#show dspfarm profile 2 Dspfarm Profile Configuration Profile ID = 2, Service = TRANSCODING, Resource ID = 2 Profile Description : Profile Service Mode : Non Secure Profile Admin State : UP Profile Operation State : ACTIVE Application : SCCP Status : ASSOCIATED Resource Provider : FLEX_DSPRM Status : UP Number of Resource Configured : 4 Number of Resource Available : 4 Codec Configuration Codec : g711ulaw, Maximum Packetization Period : 30 Codec : g711alaw, Maximum Packetization Period : 30 Codec : g729ar8, Maximum Packetization Period : 60 Codec : g729abr8, Maximum Packetization Period : 60 On Jul 6, 2010, at 10:23 AM, Graham Hopkins wrote: You'll need to do it at BR2 - if you do it at HQ/BR1 it will be G.711 across the WAN. Graham On 6 Jul 2010, at 17:41, Mark Holloway wrote: If calls should complete using G.729 from HQ/BR1 to CUE on BR2 which is G.711u, can the transcoding be configured on the BR2 router locally or does it need to happen via the originating party's transcoding resources in UCM? ___ 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] Frame-relay fragment question
Bo, When you change the frame-relay CIR settings, your fragment size will also change if you want to stay in sync with the 95% PVC best practice. Page 3-27 of the QoS SRND: Fragment Size in Bytes = (PVC Speed in kbps * Maximum Allowed Jitter in ms) / 8 If your PVC is 95% then the calculation listed above will result in a modified fragment value of 456. Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.commailto:david.ra...@kroll.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Bo Gao Sent: Tuesday, June 29, 2010 9:17 AM To: OSL Subject: [OSL | CCIE_Voice] Frame-relay fragment question HQ-BR1 bandwidth is 384K, I have the following config: map-class frame-relay AutoQoS-FR-Se0/0-201 frame-relay cir 384000 frame-relay bc 3840 frame-relay be 0 frame-relay mincir 384000 frame-relay fragment 480 service-policy output AutoQoS-Policy-Trust If I were to change the cir to 95% based on the QoS SNRD Then I would have: map-class frame-relay AutoQoS-FR-Se0/0-201 frame-relay cir 364800 frame-relay bc 3648 frame-relay be 0 frame-relay mincir 34800 frame-relay fragment 480 service-policy output AutoQoS-Policy-Trust Question: Should I also change the frame-realy fragment from 480 to 456? Why? Thank you! Bo ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Frame-relay fragment question
Graham – But if you modify the CIR, it would seem to affect the calculation used for the fragment size? Fragment Size in Bytes = (PVC Speed in kbps * Maximum Allowed Jitter in ms) / 8 Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.commailto:david.ra...@kroll.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Graham Hopkins Sent: Tuesday, June 29, 2010 9:24 AM To: Bo Gao Cc: OSL Subject: Re: [OSL | CCIE_Voice] Frame-relay fragment question I think not, the fragment size is related to the amount of data that can be placed on the wire in 10 ms which relates to line speed not CIR Graham Hopkins On 29 Jun 2010, at 15:17, Bo Gao bga...@gmail.commailto:bga...@gmail.com wrote: HQ-BR1 bandwidth is 384K, I have the following config: map-class frame-relay AutoQoS-FR-Se0/0-201 frame-relay cir 384000 frame-relay bc 3840 frame-relay be 0 frame-relay mincir 384000 frame-relay fragment 480 service-policy output AutoQoS-Policy-Trust If I were to change the cir to 95% based on the QoS SNRD Then I would have: map-class frame-relay AutoQoS-FR-Se0/0-201 frame-relay cir 364800 frame-relay bc 3648 frame-relay be 0 frame-relay mincir 34800 frame-relay fragment 480 service-policy output AutoQoS-Policy-Trust Question: Should I also change the frame-realy fragment from 480 to 456? Why? Thank you! Bo ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Frame-relay fragment question
I guess that makes sense. You're not actually making the link slower, so the fragment size wouldn't change. We'd only need to change the minCIR, CIR, and bc? Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.commailto:david.ra...@kroll.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roger Källberg Sent: Tuesday, June 29, 2010 9:33 AM To: Bo Gao; OSL Subject: Re: [OSL | CCIE_Voice] Frame-relay fragment question You shouldn't change the fragment size. Reason being that you want the fragment to be of a size that would give you a 10ms transmit delay in the event of congestion. Brgds, Roger Källberg CCIE #26199 (Voice) Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ Direkt: +46108787498 Växel: +46108787400 roger.kallb...@cygate.semailto:roger.kallb...@cygate.se Från: Bo Gao [bga...@gmail.com] Skickat: den 29 juni 2010 16:17 Till: OSL Ämne: [OSL | CCIE_Voice] Frame-relay fragment question HQ-BR1 bandwidth is 384K, I have the following config: map-class frame-relay AutoQoS-FR-Se0/0-201 frame-relay cir 384000 frame-relay bc 3840 frame-relay be 0 frame-relay mincir 384000 frame-relay fragment 480 service-policy output AutoQoS-Policy-Trust If I were to change the cir to 95% based on the QoS SNRD Then I would have: map-class frame-relay AutoQoS-FR-Se0/0-201 frame-relay cir 364800 frame-relay bc 3648 frame-relay be 0 frame-relay mincir 34800 frame-relay fragment 480 service-policy output AutoQoS-Policy-Trust Question: Should I also change the frame-realy fragment from 480 to 456? Why? Thank you! Bo ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Music on Hold
CUCME does not support intercluster multicast MOH. Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.commailto:david.ra...@kroll.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Afzal Bhutta Sent: Monday, June 28, 2010 8:36 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Music on Hold Dear Study Friends, I have issue with MOH from HQ to BR1 when I hold for MOH .My MOH is working fine for PSTN user and vice versa also working fine with in the sites. For some reason when I call from HQ to SIteB or BR1 it does not working for some reason.Please can some one puts light on it what would be the issue. Thanks for support ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Music on Hold
I was thinking in terms of IP Expert's lab setup. Afzal clarified that he was talking CUCM only so my statement does not apply in this particular scenario. Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.commailto:david.ra...@kroll.com From: Jason Aarons (US) [mailto:jason.aar...@us.didata.com] Sent: Monday, June 28, 2010 1:08 PM To: Berry, Matthew J.; Afzal Bhutta; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Music on Hold How did you infer he's using multicast and/or CME? I must be missing something. Check IP Voice Media Streaming App is enabled for G729, check regions on phones versus gateway (and/or Common Device Configuration/Device Pool). I'm guessing gateway is G711 and phones are G729 and you IP Voice Media Streaming App is only setup for G711. Just a guess. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Berry, Matthew J. Sent: Monday, June 28, 2010 11:34 AM To: Afzal Bhutta; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Music on Hold CUCME does not support intercluster multicast MOH. Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.commailto:david.ra...@kroll.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Afzal Bhutta Sent: Monday, June 28, 2010 8:36 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Music on Hold Dear Study Friends, I have issue with MOH from HQ to BR1 when I hold for MOH .My MOH is working fine for PSTN user and vice versa also working fine with in the sites. For some reason when I call from HQ to SIteB or BR1 it does not working for some reason.Please can some one puts light on it what would be the issue. Thanks for support Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP phones for CME
Make you you have bound the SIP media and control to an interface under voice service voip / sip - Sent from my Blackberry From: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com To: naoufal.kerbo...@cbi.ma naoufal.kerbo...@cbi.ma; osl osl ccie_voice@onlinestudylist.com Sent: Mon Jun 21 07:43:36 2010 Subject: Re: [OSL | CCIE_Voice] SIP phones for CME Are you working on your own gear? If so check that your phones have the correct fw hth Date: Mon, 21 Jun 2010 13:19:36 +0100 From: naoufal.kerbo...@cbi.ma To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] SIP phones for CME Hi guys, I'm working on lab9 Vol2, and I have 7961 phones registred to SIP CME, but every time the phones unregistred and registred again. Any Ideas? 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
Re: [OSL | CCIE_Voice] Connected number display
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.commailto: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.commailto:voip.ccieci...@gmail.com To: gorr...@hotmail.commailto:gorr...@hotmail.com CC: siddas...@gmail.commailto:siddas...@gmail.com; ccie_voice@onlinestudylist.commailto: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.commailto: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.commailto:siddas...@gmail.com To: voip.ccieci...@gmail.commailto:voip.ccieci...@gmail.com CC: gorr...@hotmail.commailto:gorr...@hotmail.com; ccie_voice@onlinestudylist.commailto: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.commailto: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.commailto:siddas...@gmail.com To: gorr...@hotmail.commailto:gorr...@hotmail.com CC: voip.ccieci...@gmail.commailto:voip.ccieci...@gmail.com; ccie_voice@onlinestudylist.commailto: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.commailto:voip.ccieci...@gmail.com To: ccie_voice@onlinestudylist.commailto: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
[OSL | CCIE_Voice] San Jose
Anyone in SJC this week for the OWLE? Shoot me an email if you want to grab a drink tonight. - Sent from my Blackberry ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCIE Voice #26244
Congratulations, Ash! Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.commailto:david.ra...@kroll.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ashar Siddiqui Sent: Friday, June 18, 2010 1:46 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CCIE Voice #26244 Hello all, I went to Brussels yesterday and just an hour before learned that I am now officially CCIE Voice. It was my 2nd attempt but it was worth it. I learned a lot from my first attempt and it helped me build a better strategy for the 2nd. I am thankful to this wonderful list and IPExpert material which I used. Special thanks to Amy Ryan for her help whenever I needed. I am also grateful to my Study Partner Iwan Hoogendoorn, a triple CCIE and I was so lucky to have him as Study partner. I will never forget the way he use to make daily schedules and strictly made me follow those otherwise I am a lazy man..this number is for you Iwan! Few take home points for all those who will be making an attempt in coming days: 1 - Read the lab CAREFULLY (I made it Caps for a reason)..every word in a question is there for a reason! 2 - Do not rush! the mistakes you will make in first one hour will haunt you in the entire lab (unless you are lucky to figure out what went wrong) 3 - Do not spend too much time if something is not working - you can always come back to it. 4 - Note down sections and task which you are working and cross them as soon as you have completed it 5 - Call routing - This is how I did it, not necessarily helpful for you, I did call routing on a page first as what I am going to do at RL level, Pattern level etc..I configured everything first and then tested it one by one..took me 30 minutes to finish call routing 6 - Test everything you have done at least twice and as if it was configured by someone else and you are the proctor..I found one mistake while doing my 2nd check 7 - Save your config often, make sure before you leave that all gateways are up and registered to CUCM. I joined this list for my CCIE studies when I started my CCIE journey back in December 2009 but now I have decided to stick with it as I won't find such a nice bunch of people anywhere.. N.B: Above all, I loved my number..Digit '4' is my lucky number and Cisco made sure that I have enough of them.. :) Thank you all. It's party time now ;) Ashar Siddiqui CCIE#26244 (Voice) ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 2 Lab 4 Section 3.1 - Mobile Connect Question
That worked! Thanks! Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.commailto:david.ra...@kroll.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Peter Farkas Sent: Wednesday, May 26, 2010 1:49 AM To: Matthew Berry; OSL Group Subject: Re: [OSL | CCIE_Voice] Vol 2 Lab 4 Section 3.1 - Mobile Connect Question Display ID of RDP's DN is missing. When shared line is created then only the Alerting Name is copied from the line. Go to the DN Configuration of 5002 and select the RDP from Associated Devices list and use Edit Line Appaearance button to modify. - Original Message - From: Matthew Berrymailto:ciscovoiceg...@gmail.com To: OSL Groupmailto:ccie_voice@onlinestudylist.com Sent: Wednesday, May 26, 2010 3:11 AM Subject: [OSL | CCIE_Voice] Vol 2 Lab 4 Section 3.1 - Mobile Connect Question Fellow nerds, I am battling a single number reach (i.e. Mobile Connect) question on Lab 4. Question 3.1 says the call should appear to BR1 Phone 2 as if it is actually coming from HQ Phone 2 directly (Calling Name and Number). When I call in from the PSTN phone to BR1 Phone 2, the display on BR1 Phone 2 shows 5002. The calling number is represented just fine. However, I cannot get the calling nmae to be presented on the display. I have tinkered around with the partial/complete match and significant digits parameters under the mobility section of the Call Manager service parameters but nothing has changed. Any ideas? -- Matthew Berry A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written Vitals: GVoice: +1.612.424.5044 Gmail: ciscovoiceg...@gmail.commailto:ciscovoiceg...@gmail.com Skype: ciscovoiceguru Twitter: ciscovoiceguru Cert Stats: Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Show Gatekeeper Calls - Odd Issue
All - I had an issue last night on Vol 2 Lab 2. I am sending calls from HQ (Region = HQ) to BR2 over my H.225 trunk (Region = GK). Region setting between HQ and GK specifies G.729. I have a transcoder registered on the BR2 router. When I call across the gatekeeper, my endpoints show G.729, but show gatekeeper calls shows 128kbps. Extremely odd. Does anyone have insight into this? Thanks! Matthew Berry, CCVP, Sr. Unified Communications Engineer Kroll | 9023 Columbine Road, Eden Prairie, MN 55347 Single Number Reach +1 952 516 3748 | Fax +1 952 516 3646 | mjbe...@kroll.commailto:david.ra...@kroll.com www.krollontrack.comhttp://www.krollbackgroundscreening.com/ | www.kroll.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Show Gatekeeper Calls - Odd Issue
I did enable BRQ as a troubleshooting method. After applying the parameter and resetting my devices, the problem still existed. Here is the dial-peer on my BR2-RTR: dial-peer voice 30 voip destination-pattern [15]...$ session target ras incoming called-number . tech-prefix 1# no vad Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.com -Original Message- From: ccieid1ot [mailto:ccieid...@gmail.com] Sent: Friday, May 21, 2010 10:23 AM To: Berry, Matthew J. Cc: CCIE Voice OSL (ccie_voice@onlinestudylist.com) Subject: Re: [OSL | CCIE_Voice] Show Gatekeeper Calls - Odd Issue BRQ enabled? Hardcode G.729 on incoming dial-peer? On Fri, May 21, 2010 at 10:21 AM, Berry, Matthew J. mjbe...@krollontrack.com wrote: All - I had an issue last night on Vol 2 Lab 2. I am sending calls from HQ (Region = HQ) to BR2 over my H.225 trunk (Region = GK). Region setting between HQ and GK specifies G.729. I have a transcoder registered on the BR2 router. When I call across the gatekeeper, my endpoints show G.729, but show gatekeeper calls shows 128kbps. Extremely odd. Does anyone have insight into this? Thanks! Matthew Berry, CCVP, Sr. Unified Communications Engineer Kroll | 9023 Columbine Road, Eden Prairie, MN 55347 Single Number Reach +1 952 516 3748 | Fax +1 952 516 3646 | mjbe...@kroll.com www.krollontrack.com | www.kroll.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] Show Gatekeeper Calls - Odd Issue
That's great to know. I burned a few hours last night on Proctor trying to get this to work. Hopefully we won't be asked a question like that on the lab. According to my understanding, then, we cannot technically complete and get points for question 5.1 since it requires you to produce the show gatekeeper calls output listed in the question. Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.commailto:david.ra...@kroll.com From: Roger Källberg [mailto:roger.kallb...@cygate.se] Sent: Friday, May 21, 2010 10:44 AM To: Berry, Matthew J.; CCIE Voice OSL (ccie_voice@onlinestudylist.com) Subject: RE: [OSL | CCIE_Voice] Show Gatekeeper Calls - Odd Issue Also this, http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg16188.html Roger Källberg Unified Communication Consultant Cygate AB From: Berry, Matthew J. [mailto:mjbe...@krollontrack.com] Sent: den 21 maj 2010 17:21 To: CCIE Voice OSL (ccie_voice@onlinestudylist.com) Subject: [OSL | CCIE_Voice] Show Gatekeeper Calls - Odd Issue All - I had an issue last night on Vol 2 Lab 2. I am sending calls from HQ (Region = HQ) to BR2 over my H.225 trunk (Region = GK). Region setting between HQ and GK specifies G.729. I have a transcoder registered on the BR2 router. When I call across the gatekeeper, my endpoints show G.729, but show gatekeeper calls shows 128kbps. Extremely odd. Does anyone have insight into this? Thanks! Matthew Berry, CCVP, Sr. Unified Communications Engineer Kroll | 9023 Columbine Road, Eden Prairie, MN 55347 Single Number Reach +1 952 516 3748 | Fax +1 952 516 3646 | mjbe...@kroll.commailto:david.ra...@kroll.com www.krollontrack.comhttp://www.krollbackgroundscreening.com/ | www.kroll.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Show Gatekeeper Calls - Odd Issue
The workaround is to set both service parameters to g729 (listed below). Really? · Intraregion Audio Codec Default Required Field · Interregion Audio Codec Default Required Field If we were required to show that output, we'd need to configure intraregion to g729 and then manually define g711-to-g711 for within each region. That's just crazy talk. :) CSCsl74701 Bug Details ARQ requests 1280 when no regions are defined to use g711 Symptom: ARQ sent to gatekeeper requests bandwidth for a g711 call (1280) even though only g729 is configured in all of the regions. Conditions: In a call routed to a GK controlled ICT and all regions are configured for g729, the originating CCM requests 160 in the ARQ to the gatekeeper. When the h225 setup arrives at the terminating CCM, an ARQ is sent to the gatekeeper requesting 1280. This is because the IntraAudioRegionDefault and InterAudioRegionDefault service paramater settings are included in the calculation for the maximum bandwidth request. Callmanager default setting for IntraAudioRegionDefault is g711. It should check region pair before applying default if there is nothing matched. Workaround: Set both service parameters to g729, or increase zone bandwidth setting on the gatekeeper Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.commailto:david.ra...@kroll.com From: Roger Källberg [mailto:roger.kallb...@cygate.se] Sent: Friday, May 21, 2010 10:44 AM To: Berry, Matthew J.; CCIE Voice OSL (ccie_voice@onlinestudylist.com) Subject: RE: [OSL | CCIE_Voice] Show Gatekeeper Calls - Odd Issue Also this, http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg16188.html Roger Källberg Unified Communication Consultant Cygate AB From: Berry, Matthew J. [mailto:mjbe...@krollontrack.com] Sent: den 21 maj 2010 17:21 To: CCIE Voice OSL (ccie_voice@onlinestudylist.com) Subject: [OSL | CCIE_Voice] Show Gatekeeper Calls - Odd Issue All - I had an issue last night on Vol 2 Lab 2. I am sending calls from HQ (Region = HQ) to BR2 over my H.225 trunk (Region = GK). Region setting between HQ and GK specifies G.729. I have a transcoder registered on the BR2 router. When I call across the gatekeeper, my endpoints show G.729, but show gatekeeper calls shows 128kbps. Extremely odd. Does anyone have insight into this? Thanks! Matthew Berry, CCVP, Sr. Unified Communications Engineer Kroll | 9023 Columbine Road, Eden Prairie, MN 55347 Single Number Reach +1 952 516 3748 | Fax +1 952 516 3646 | mjbe...@kroll.commailto:david.ra...@kroll.com www.krollontrack.comhttp://www.krollbackgroundscreening.com/ | www.kroll.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Problem with CUPC (Lab 13 Vol1)
Without LDAP integration, you will not be able to do directory search. To enable chat with LDAP, you must first send a message from the IP phone presence client to the CUPC. You will then be able to add the contact and chat. Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.com -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of kerboute kerboute Sent: Friday, May 21, 2010 2:36 PM To: CCIE Voice Maillist Subject: [OSL | CCIE_Voice] Problem with CUPC (Lab 13 Vol1) Hi guys, Yesterday I worked on the the lab 13 Vol1 for presence, And I cannot got CUPC work properly, the CUPC connect to presence but with Limited state and I cannot get the directory search and chat ... Any Ideas? Regards Naoufal ___ 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] problem lab 5A 5.4
Do you have the appropriate called party transformation CSS configured on your gateway? Make sure you have unchecked the option to use the device pool CSS for transformations. It sounds like your transformation is not being invoked on the gateway. Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.commailto:david.ra...@kroll.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of amr gaber Sent: Tuesday, May 18, 2010 9:15 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] problem lab 5A 5.4 In this part we should configured called party transformation pattern when we dial a E164 number from missed call (+1212-394-2123) we should get 3942123 on the router to dial pstn (after match dial-peer on HQ router) But actually it doesn't work. when I debug the out in HQ router I get as there's no transformation (the dialed number match the pattern and reach the HQ reuter but without any transformation) I tried to get the required result with old way (and I can achieved the required question by using the called party transformation from the same page of Route Pattern but I need to test called transformation) I am still in this session , I need your help and idea Thanks Amr Thabt ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Trancoding resources on CUBE
Bo - You're right. I changed the max sessions value from 45 to 2. This is the new output from my show sdspfarm units: mtp-1 Device:RTR-XCODE TCP socket:[1] REGISTERED in SCCP ver 17/10 actual_stream:4 max_stream 4 IP:192.168.99.1 57105 MTP Dixieland keepalive 0 Supported codec: G711Ulaw G711Alaw G729 G729a G729ab max-mtps:1, max-streams:90, alloc-streams:4, act-streams:0 Two values to point out: MAX-STREAMS remains the same. This is the number of sdspfarm transcode sessions I specified under telephony-service. You're right, in that xocde/mtp resources are counted as multiples of two. Since I stated 45 sessions, it lists 90 streams. ALLOC-STREAMS reflects the max-sessions listed under the dspfarm profile section of my configuration. Since I entered 2 sessions, it displays 4 streams (again, multiples of two). What I need pay attention to is SESSIONS versus STREAMS. Lastly, what the heck is MTP Dixieland? (second line of the output). That's weird. Matthew Berry Digital Footprint: Twitter: ciscovoiceguru Skype: ciscovoiceguru 1st Lab Attempt: Aug 16th, 2010 From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Bo Gao [bga...@gmail.com] Sent: Friday, May 07, 2010 7:22 AM To: Matthew Berry Cc: OSL Subject: Re: [OSL | CCIE_Voice] Trancoding resources on CUBE I think the number 90 was not b/c 45 (dspfarm) + 45 (sdspfarm), you had a value of 90 because each transcoder session consists of two transcoding streams between callers using transcode. Bo On Fri, May 7, 2010 at 3:33 AM, Matthew Berry ciscovoiceg...@gmail.commailto:ciscovoiceg...@gmail.com wrote: Earlier this week, I began Vol 2 Lab 1. In this lab, I configured transcoding resources on the CUBE. These resources were registered to the gateway itself, under telephony-service. I was messing around on a router this morning and found something confusing. If I define maximum sessions under the dspfarm profile as well as sdspfarm transcode sessions under telephony-service, the values seem to be considered independent of each other. I defined 45 maximum sessions on the dspfarm profile. Hower, when I run a show sdspfarm units, I get a total of 90 max-streams. The two commands appear to be summed up in this command. Can someone explain this to me? sccp local Loopback1 sccp ccm 192.168.99.1 identifier 1 version 7.0 sccp ! sccp ccm group 1 bind interface Loopback1 associate ccm 1 priority 1 associate profile 1 register RTR-XCODE signaling dscp ef ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 45 associate application SCCP ... telephony-service sdspfarm units 1 sdspfarm transcode sessions 45 sdspfarm tag 1 RTR-XCODE max-ephones 1 max-dn 1 ip source-address 192.168.99.1 port 2000 max-conferences 8 gain -6 transfer-system full-consult create cnf-files version-stamp Jan 01 2002 00:00:00 -- Matthew Berry A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written Vitals: GVoice: +1.612.424.5044 Gmail: ciscovoiceg...@gmail.commailto:ciscovoiceg...@gmail.com Skype: ciscovoiceguru Twitter: ciscovoiceguru Cert Stats: Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Device Pool over Phone Configuration settings
Vik is absolutely right. In Chapter 9 of the CUCM SRND (9-12): Location Hub_None is a special location that is configured by default with unlimited audio and video bandwidth, and location Hub_None cannot be deleted. If the devices at a branch location are configured in the Hub_None location, none of the phone calls to or from that branch device will be subject to any call admission control. Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.com -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ryan Schwab Sent: Thursday, May 06, 2010 12:49 PM To: 'Vik Malhi' Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Device Pool over Phone Configuration settings Ok, so if a non-default (ie. NOT hub_none) Location is selected on the Device, it is chosen over the DP location? -Original Message- From: Vik Malhi [mailto:vma...@ipexpert.com] Sent: May-06-10 11:35 AM To: Ryan Schwab Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Device Pool over Phone Configuration settings NOT TRUE! If device location =hub_none dp location is used. -- Vik Malhi – CCIE #13890 Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ On May 6, 2010, at 10:27 AM, Ryan Schwab schwab...@shaw.ca wrote: It would seem logical to me that the Location and MRGL selected on the Phone configuration would override the settings in the Device Pool. However, I remember hearing that the Device pool Location always wins over the Location set on the phone itself. Is this true? If so, can anyone summarize the theory behind it? ___ 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 Lab 1 - ISDN Restart Errors
I encountered this linecode issue last week. Changing it to B8ZS fixed my problem. Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.commailto:david.ra...@kroll.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roger Källberg Sent: Thursday, May 06, 2010 2:57 PM To: Steve Denney (stdenney); ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Vol 2 Lab 1 - ISDN Restart Errors I believe it has to do with the line code used on the controller. It is set to linecode ami in the initial config, but should be linecode b8zs. Roger Källberg Unified Communication Consultant Cygate AB From: Steve Denney (stdenney) [mailto:stden...@cisco.com] Sent: den 6 maj 2010 21:27 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Vol 2 Lab 1 - ISDN Restart Errors Hi, Seeing some errors today that I haven't encountered before in any other lab...wh! :) I'm working on Vol 2 Lab 1 Question 4.1, and trying to get calls from the PSTN working into HQ. Pretty straightforward stuff, except the calls never seem to get out of the PSTN router. When dialing the HQ phone from the PSTN phone (regardless of line selected), I get the following debug isdn q931 errors from the PSTN router: May 6 23:11:09.243: ISDN Se0/3/0:23 Q931: Applying typeplan for sw-type 0xD is 0x0 0x0, Calling num 911 May 6 23:11:09.243: ISDN Se0/3/0:23 **ERROR**: CCPMSG_OutCall: fails with cause 0x22 And every 30 seconds, I see the same batch of 4 ISDN Restart messages, like this (also from the PSTN router): May 6 23:12:04.539: ISDN Se0/3/0:23 Q931: TX - RESTART pd = 8 callref = 0x Restart Indicator i = 0x87 May 6 23:12:05.539: ISDN Se0/3/0:23 Q931: TX - RESTART pd = 8 callref = 0x Restart Indicator i = 0x87 May 6 23:12:06.539: ISDN Se0/3/0:23 Q931: TX - RESTART pd = 8 callref = 0x Restart Indicator i = 0x87 May 6 23:12:07.539: ISDN Se0/3/0:23 Q931: TX - RESTART pd = 8 callref = 0x Restart Indicator i = 0x87 Show isdn status on the PSTN router looks normal for this interface: ISDN Serial0/3/0:23 interface *** Network side configuration *** dsl 1, interface ISDN Switchtype = primary-ni Layer 1 Status: ACTIVE Layer 2 Status: TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED Layer 3 Status: 0 Active Layer 3 Call(s) Active dsl 1 CCBs = 0 The Free Channel Mask: 0x8000 Number of L2 Discards = 0, L2 Session ID = 0 Attaching show run and show isdn status as well for the HQ router (the other end) just for troubleshooting completeness, but there's no indication of anything amiss, nor any debug messages at all, on the HQ router. The call never gets that far. I started this morning on Voice Pod 11 and hit this. Ryan was kind enough to move me over to Voice Pod 16, but I'm hitting the same issue here. OSL archive and Google search turned up nothing concrete, other than a general theme of it sounds like your telco / carrier has issues. :) Any ideas? Cheers and TIA, sd ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] SIP Supplementary Services
Here's my question: What kind of real-world (or lab) scenario would require disabling SIP supplementary services on an IOS gateway? Quote from CUCM SRND: SIP Refer or SIP 302 Moved Temporarily messages can be used for supplementary services such as call transfer or call forward on Unified CME or Unified CM to instruct the transferee (referee) or phone being forwarded (forwardee) to initiate a new call to the transfer-to (refer-to) target or forward-to target. No hairpinning is needed for call transfer or call forward scenarios when the SIP Refer or SIP 302 Moved Temporarily message is supported. However, supplementary-service must be disabled if there are certain extensions that have no DID mapping or if Unified CM or Unified CME does not have a dial plan to route the call to the DID in the SIP 302 Moved Temporarily message. When supplementary-service is disabled, Unified CME hairpins the calls or sends a re-invite SIP message to Unified CM to replace the media path to the new called party ID. Both signaling and media are hairpinned, even when multiple Unified CMEs are involved for further call forwards. The supplementary-service can also be disabled for transferred calls. In this case, the SIP Refer message will not be sent to Unified CM, but the transferee (referee) party and transfer-to party (refer-to target) are hairpinned. Note Supplementary services can be disabled with the command no supplementary-service sip moved-temporarily or no supplementary-service sip refer under voice service voip or dial-peer voice voip. Thanks! Matthew Berry, CCVP, Sr. Unified Communications Engineer Kroll | 9023 Columbine Road, Eden Prairie, MN 55347 Single Number Reach +1 952 516 3748 | Fax +1 952 516 3646 | mjbe...@kroll.commailto:david.ra...@kroll.com www.krollontrack.comhttp://www.krollbackgroundscreening.com/ | www.kroll.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 1 Lab 13 - CUCM + CUPS Presence
CUPC STATUS MENU IS GRAYED-OUT This usually means that CUPC failed to connect to the Presence Engine. This could be caused by: * Digest credential or Incoming ACL was not configured * Proxy domain was not configured properly * Network issue Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.commailto:david.ra...@kroll.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ShinGei Yong Sent: Friday, April 30, 2010 6:05 AM To: ccie_voice@onlinestudylist.com; ciscovoiceg...@gmail.com Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 13 - CUCM + CUPS Presence Hi Matthew, I'm encounter a very similar problem as you, which the CUPC status not shown anything. From the Show server Health from CUPC, the process halt at the Presence status, symptom is it keep connecting disconnecting. I've configured the proxy under the proxy domain, and i can login to CUPC as well. How do you resolve the issue? Thanks Shingei On Tue, Apr 27, 2010 at 3:29 PM, Angel Perez gorr...@hotmail.commailto:gorr...@hotmail.com wrote: Hi I forgot to say that: 3: At CUC go to userpasswordweb password uncheck user have to change password next login then at CUPC go to Filepreference and add web username and pass at voice mail account From: gorr...@hotmail.commailto:gorr...@hotmail.com To: ciscovoiceg...@gmail.commailto:ciscovoiceg...@gmail.com; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Date: Tue, 27 Apr 2010 07:01:58 + Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 13 - CUCM + CUPS Presence Hi: Assuming that you have taken all the neccesary steps to successfully integrate UCM and CUPS try the following: 1: Sometimes you have to restart the cups appliance if you have changed name to ip address 2: This is the normal situation, you would need to add the IPPM service to the hard phone associated to CUPC then add the users from the service menu at hard phone, then you would see other users and it presence. 3: At CUC CoS, check allow to use imap and allow to acces messages bodies, sometimes MWI notification on CUPC takes 30-40 sec hth Date: Mon, 26 Apr 2010 12:30:57 -0500 From: ciscovoiceg...@gmail.commailto:ciscovoiceg...@gmail.com To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Vol 1 Lab 13 - CUCM + CUPS Presence All- I've been having a difficult time tonight trying to get CUPS Presence configured and working correctly. I followed the Proctor Guide, referencing Vik's Vol 1 Walkthroughs and HouTong Luo's Deploying Cisco Unified Presence book. Here's the end result. I can control my phones through the CUPC client. When calls ring inbound, I can divert them to voicemail via the client. All the CUPS services are up and running (although, I did have to reboot the server to get AXL to remain up, much like my CUC and MWI email from late last week). Several things, I cannot get to work: 1. Client status - My menu bars are mostly grayed out. When I run the Presence Viewer on the CUPS, it cannot see my presence, even though I am logged in with CUPC. 2. CUPC directory lookup - I cannot lookup and find BR1-Phone2 or HQ-Phone2 in the directory. It is empty. HuoTong mentioned that directory lookup was impossible without an AD integration. Is this true? 3. Voicemail notification - I went into CUC and enabled Allow Users to Use Unified Client to Access Voice Mail under CoS. My phones can interact with MWI but not the CUPC client. When I run the troubleshooters, everything comes back green. I have checked line associate, user profiles, SIP trunk, SIP trunk security profile, etc. Any ideas? Error! Filename not specified. -- Matthew Berry A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written Vitals: GVoice: +1.612.424.5044 Gmail: ciscovoiceg...@gmail.commailto:ciscovoiceg...@gmail.com Skype: ciscovoiceguru Twitter: ciscovoiceguru Cert Stats: Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 Hotmail: Free, trusted and rich email service. Get it now.https://signup.live.com/signup.aspx?id=60969 Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now.https://signup.live.com/signup.aspx?id=60969 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Unable to open .aef file on CRS editor
Jeff - Use the username and password that is used to login to AppAdmin. Pay attention! The login is case-sensitive. If AppAdam/CUCM sees your login ID as JCotter, you better enter it exactly (jcotter will not work). There should be a button on the login screen to login anonymously. Really the only different is that you won't be able to load repository scripts from the Editor and reactive debugging won't work. Let me know how it goes. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of vccie2010 Sent: Thursday, April 29, 2010 12:57 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Unable to open .aef file on CRS editor I am trying to open .aef file on CRS edittor, when I open the file, CRS Editor prompts for user name, pass and IP for server. I tried with annonymous and other regular usrname but no luck.am I missing somehing hereI remeber I was able to login offline on CRS editor with u/p - annonymous/ annonymous on UCCX 4.0 thx for your help ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Unable to open .aef file on CRS editor
If using PL, did you try admin and c1sc0123? From: vccie2010 [mailto:vccie2...@gmail.com] Sent: Thursday, April 29, 2010 1:20 PM To: Berry, Matthew J. Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Unable to open .aef file on CRS editor no luck tired with AppAdmin and Annonymous On Thu, Apr 29, 2010 at 11:02 AM, Berry, Matthew J. mjbe...@krollontrack.commailto:mjbe...@krollontrack.com wrote: Jeff - Use the username and password that is used to login to AppAdmin. Pay attention! The login is case-sensitive. If AppAdam/CUCM sees your login ID as JCotter, you better enter it exactly (jcotter will not work). There should be a button on the login screen to login anonymously. Really the only different is that you won't be able to load repository scripts from the Editor and reactive debugging won't work. Let me know how it goes. From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of vccie2010 Sent: Thursday, April 29, 2010 12:57 PM To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Unable to open .aef file on CRS editor I am trying to open .aef file on CRS edittor, when I open the file, CRS Editor prompts for user name, pass and IP for server. I tried with annonymous and other regular usrname but no luck.am I missing somehing hereI remeber I was able to login offline on CRS editor with u/p - annonymous/ annonymous on UCCX 4.0 thx for your help ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Problem in LAB 3A vol 1
Amy - So you recommend that we rely purely on CUCM to do the firmware conversion for the CUCME phones? I know there are varying opinions about whether this is a good option or not. Most of it, in my opinion, stems from the (legitimate?) fear that the lab will specifically ask us to change the firmware on the phones to a specific version. Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.commailto:david.ra...@kroll.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Amy Ryan Sent: Tuesday, April 20, 2010 6:16 PM To: amr gaber; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Problem in LAB 3A vol 1 It looks as if the current firmware load on your 7960 device is SCCP. It is a painful process to download the SIP firmware from the CCME router. It is recommended that you first add the mac-address of your 7960 phone to the UCM as a SIP endpoint. Then in your dhcp pool, use the CUCM as the option 150 temporarily. Once the SIP firmware download is completed, you can switch the option 150 back to the local BR2 tftp server. HTH, Amy --- Amy Ryan - CCIE #24677 (Voice) Technical Instructor - IPexpert, Inc. Mailto: ar...@ipexpert.com Telephone: +1.810.326.1444 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat eFax: +1.810.454.0130 IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ From: amr gaber amrga...@gmail.com Date: Wed, 21 Apr 2010 01:55:51 +0300 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Problem in LAB 3A vol 1 Dear, I working on vol 1 lab 3A Part 3.5 for configuration page 149-151 for verification page 153-156 the problem I can't get the BR2 PHONE on fa 0/3/1 to register when I verify with Sh flash | i .cnf I can see the SIP file for the phone more details when I debug tftp event I get:- Apr 20 21:21:16.293: TFTP: Looking for CTLSEP00131A1E579D.tlv Apr 20 21:21:16.361: TFTP: Looking for SEP00131A1E579D.cnf.xml Apr 20 21:21:16.361: TFTP: Opened system:/its/XMLDefault7960. cnf.xml, fd 7, size 971 for process 339 Apr 20 21:21:16.365: TFTP: Finished system:/its/XMLDefault7960.cnf.xml, time 00:00:00 for process 339 It seems that no looking for SIP00131A1E579D.cnf (which was created and appears in flash) also I get this message in terminal monitor Apr 20 21:26:56.730: %IPPHONE-6-REG_ALARM: 24: Name=SEP00131A1E579D Load=8.0(5.0) Last=Phone-Reg-Rej please advise ___ 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] Vol2 Lab 4 - Presense - CUPS not working
Great post, Otto. I miss chatting with you! I'll need to keep my eyes open for you on Skype. Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.commailto:david.ra...@kroll.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Otto Sanchez Sent: Wednesday, March 03, 2010 6:03 AM To: Wael Agina Cc: OSL Group Subject: Re: [OSL | CCIE_Voice] Vol2 Lab 4 - Presense - CUPS not working Hi Wael, Please follow the guidelines in this blog to control your phone cti from cupc and let us know the results: http://blog.ipexpert.com/cti-phone-control-from-cupc/#more-2272 Thanks, On Wed, Mar 3, 2010 at 4:24 AM, Wael Agina waelag...@gmail.commailto:waelag...@gmail.com wrote: Dear All, I've integerate the cups - 10.10.210.12 with cucm. I assigned the user gwashington as per PG to line 5002 of HQ PH 2 / My CIPC phone. I am login using my CUPC client on my machine to the CUPS 10.10.210.12 , however it is not working fine. I can login, but not monitoring or dialing via phone hq ph 2 as supposed. User gwashington has device associated hq phe 2 and primary line 5002. Any idea ? -- Thanks and Best Regards, Wael Agina ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Converting CUE to Integrate with UCM
Jeff - Did you ever find out the answer to your question? I'm curious. Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.commailto:david.ra...@kroll.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Cotter Sent: Monday, March 22, 2010 6:12 PM To: osl osl Subject: [OSL | CCIE_Voice] Converting CUE to Integrate with UCM Is the only requirement to go from CME integration to UCM to load the proper license file? This is my companies equipment not proctor labs. I would like to be able to move back and forth similar to proctor labs but am unsure it is as easy as just loading the proper license file. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] [cisco-voip] Plz help, anyone knows the order of operation of Calling party transformation patterns in CUCM 7.0.1?
Jeremy - It's important to remember that translation pattern modify the number whereas route patterns and route lists do not. Once the modification takes place, as a call passes from an internal IP phone toward the gateway, the transformations at each point can be overridden. The called/calling parting transformations applied at the CUCM gateway configuration will always override the settings before it. Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.commailto:david.ra...@kroll.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roger Källberg Sent: Thursday, April 15, 2010 2:02 AM To: jeremy co; le...@uoguelph.ca Cc: ccie_voice@onlinestudylist.com; cisco-v...@puck.nether.net Subject: Re: [OSL | CCIE_Voice] [cisco-voip] Plz help, anyone knows the order of operation of Calling party transformation patterns in CUCM 7.0.1? Hi Jeremy, You got this answer from Otto Sanchez on your other post on the same topic. Hello Jeremy, I think that an important fact to know is that cg/cd xform patterns get matched at the time the rp is being hit, so cg/cd xforms will override the transformations you perform at the rp level, If you configure the EPNM in a TP, the cg number getting to the rp will be the globalized number, right?, in this case, a cg xform pattern will be matched only if it corresponds to that EPNM, Please let me know if this clears up things a little bit for you, I would say that the answer you got is pretty strait forward, but if not test it out in all sort of ways in your lab and I think it will click for you. Best regards Roger Källberg Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ Från: jeremy co [jeremy.coo...@gmail.com] Skickat: den 15 april 2010 04:22 Till: le...@uoguelph.ca Kopia: ccie_voice@onlinestudylist.com; cisco-v...@puck.nether.net Ämne: Re: [OSL | CCIE_Voice] [cisco-voip] Plz help, anyone knows the order of operation of Calling party transformation patterns in CUCM 7.0.1? well, I know this order but CngPTP break these rules. Check out this scenario: callTPRP--- SLRG-GW ,check EPNM on TP and RP ,put CngPTP on GW.it overrides. ,check EPNM only on RP ,put CngPTP on GW.it does not override. ,check EPNM only on TP ,put CngPTP on GW.it overrides. How these mess works with CngPTP? Jeremy On Thu, Apr 15, 2010 at 12:13 PM, le...@uoguelph.camailto:le...@uoguelph.ca wrote: The farther down the chain you go the override happens. So: gateway overrides RL RL overrides RG RG overrides RP RP overrides phone Pretty sure about this, but I'm sure someone will chime in. Lelio Fulgenzi, Senior Analyst Computing Communications University of Guelph 519-824-4120 x56354 ...sent from my iPod - please pardon my fat fingers ;) [XKJ2000] On 2010-04-14, at 10:09 PM, jeremy co jeremy.coo...@gmail.commailto:jeremy.coo...@gmail.com wrote: Hi, Anyone knows how calling party transformation pattern order of operation in CUCM 7.0.1? I just got confused and docs are pretty weak in this area. It should override RP,RG,RL configuration. But apparently it would not! What is the relationship of checking EPNM on TPRP,GW and CngPTP ? I really appreciate any comment on this. Any doc that explain this relationship? Cheers Jeremy ___ cisco-voip mailing list cisco-v...@puck.nether.netmailto:cisco-v...@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Problem in LAB 3A vol 1
Thanks, Amy. Good tips! Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.commailto:david.ra...@kroll.com From: Amy Ryan [mailto:ar...@ipexpert.com] Sent: Wednesday, April 21, 2010 11:19 AM To: Berry, Matthew J.; amr gaber; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Problem in LAB 3A vol 1 Matthew, I am not saying to purely rely on the CUCM, just making a recommendation. Your fear is valid it is entirely possible the proctor could require you to use the a specific version that may either be located only local to the CUCME router or not, as is the same with many other examples we could toss about. So yes, in your journey it is good to be familiar with this process, but as in this case, it is equally valuable to know other methods that can save time when applicable. For what it is worth, when using the process via CUCME, this debug will be your best companion. debug tftp events You can also ensure the cnf.xml files are created by using the following two commands. show voice register tftp show telephony-service tftp Thank you, Amy --- Amy Ryan - CCIE #24677 (Voice) Technical Instructor - IPexpert, Inc. Mailto: ar...@ipexpert.com Telephone: +1.810.326.1444 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat eFax: +1.810.454.0130 IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ From: Berry, Matthew J. mjbe...@krollontrack.com Date: Wed, 21 Apr 2010 10:09:25 -0500 To: Amy Ryan ar...@ipexpert.com, amr gaber amrga...@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Problem in LAB 3A vol 1 Amy - So you recommend that we rely purely on CUCM to do the firmware conversion for the CUCME phones? I know there are varying opinions about whether this is a good option or not. Most of it, in my opinion, stems from the (legitimate?) fear that the lab will specifically ask us to change the firmware on the phones to a specific version. Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.com mailto:david.ra...@kroll.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Amy Ryan Sent: Tuesday, April 20, 2010 6:16 PM To: amr gaber; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Problem in LAB 3A vol 1 It looks as if the current firmware load on your 7960 device is SCCP. It is a painful process to download the SIP firmware from the CCME router. It is recommended that you first add the mac-address of your 7960 phone to the UCM as a SIP endpoint. Then in your dhcp pool, use the CUCM as the option 150 temporarily. Once the SIP firmware download is completed, you can switch the option 150 back to the local BR2 tftp server. HTH, Amy --- Amy Ryan - CCIE #24677 (Voice) Technical Instructor - IPexpert, Inc. Mailto: ar...@ipexpert.com Telephone: +1.810.326.1444 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat eFax: +1.810.454.0130 IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ From: amr gaber amrga...@gmail.com Date: Wed, 21 Apr 2010 01:55:51 +0300 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Problem in LAB 3A vol 1 Dear, I working on vol 1 lab 3A Part 3.5 for configuration page 149-151 for verification page 153-156 the problem I can't get the BR2 PHONE on fa 0/3/1 to register when I verify with Sh flash | i .cnf I can see the SIP file for the phone more details when I debug tftp event I get:- Apr 20 21:21:16.293: TFTP: Looking for CTLSEP00131A1E579D.tlv Apr 20 21:21:16.361: TFTP: Looking for SEP00131A1E579D.cnf.xml Apr 20 21:21:16.361: TFTP: Opened system:/its/XMLDefault7960. cnf.xml, fd 7, size 971 for process 339 Apr 20 21:21:16.365: TFTP: Finished system:/its/XMLDefault7960.cnf.xml, time 00:00:00 for process 339 It seems that no looking for SIP00131A1E579D.cnf (which was created and appears in flash) also I get this message in terminal monitor Apr 20 21:26:56.730: %IPPHONE-6-REG_ALARM: 24: Name
Re: [OSL | CCIE_Voice] CUE integration with multiple CMEs
I read the same thing the other day. You can actually configure multiple CUCME sites with a single CUE. You will need to setup the CUCME hosting the CUE as an MWI relay server. To Configure the SIP MWI Server (Multiple CUCME Routers) o Go into the SIP user-agent config and configure an IP address and port for the SIP MWI server. * expires = (optional) Subscription expiration time, in seconds. The default is 3600. * transport tcp = The default setting * transport udp = Allows you to integrate with the SIP MWI client * port = Used to specify TCP port for the SIP MWI server. The default SIP port is 5060. * unsolicited = Allows sending SIP NOTIFY for MWIs without the need to send as SUBSCRIBE from the CUCME router. Example: sip-ua mwi-server ipv4:10.10.210.15 ! telephony-service mwi reg-e164 mwi relay ! Optional, enables router to relay MWI info to other CUCME Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.commailto:david.ra...@kroll.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Cotter Sent: Wednesday, April 21, 2010 1:40 PM To: osl osl Subject: [OSL | CCIE_Voice] CUE integration with multiple CMEs Looking for some clarification on support for multiple CME sites with a single CUE module and provide MWI notification to remote sites. Release notes for 3.1 indicate support for integration to multiple CME however admin guide and design guides state the following: Restrictions for Integrating Cisco CME with Cisco Unity Express Cisco Unity Express cannot provide voice-mail services across Cisco CME routers. Cisco Unity Express can provide voice mail services only for phones on its host Cisco CME router. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 2 Lab 10 Q4.6 - VoiceView and CUCM - Authentication Error
Sergio - Did you ever figure this out? Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.commailto:david.ra...@kroll.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Sergio Polizer Sent: Wednesday, April 14, 2010 8:28 AM To: amccar...@cciequest.com; ar...@ipexpert.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Vol 2 Lab 10 Q4.6 - VoiceView and CUCM - Authentication Error Hello Amy, Amp, They were the steps that I had done. I can see and manage my messages. I just can not play them. I think that there is something related to the authentication link. I remember when integrating w/ CUCME I could listening my messages after I added the following two authentication link: CUCME: Authentication service URL: http://CUE-hostname/voiceview/authentication/authenticate.do CUE: Authentication Fallback Server URL: http://CUCME /CCMCIP/authenticate.asp But when integrating with CUCM, the authentication link is already configured at Enterprise Parameters: http://CUCM:8080/ccmcip/authenticate.jsp. I tried to change it as the same as we done when integrating c/ CUCME but there is no place to configure the fallback authentication link at CUE and not worked. I'll clean all my configurations and try again. I'll let you know the results. If you have any other suggestion, please let me know. Thank you. Sergio. Date: Tue, 13 Apr 2010 03:45:42 -0400 From: amccar...@cciequest.com To: ar...@ipexpert.com CC: spoli...@hotmail.com; moataz_m...@yahoo.com; mlin...@conet.de; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Vol 2 Lab 10 Q4.6 - VoiceView and CUCM - Authentication Error Thanks for that info Amy. I found a link that should be helpful. http://www.cisco.com/en/US/products/sw/voicesw/ps5520/products_tech_note09186a008067cb94.shtml Amp Quoting Amy Ryan ar...@ipexpert.com: Hello Sergio, Thank you for bringing to our attention that the solution was missing. This has now been added and the update should be available for download shortly. In this instance, there should not be an issue with the license and IVR sessions are not required as part of the license. It appears below that you have the correct license. (CCM) Please ensure you have completed the following steps: In Cisco Unity Express 1. Enabled Voiceview In CUCM 1. Added Voiceview as a phone service (Service url: http://10.10.202.2/voiceview/common/login.do) 2. Applied service to the appropriate phone and reset phone 3. Associated phone as a controlled device with the ³cue² Application User (just as you have for cti ports and cti route point) Once the above steps are completed, you should be able to press the Services button and see the menu option for Voiceview (or the name you gave the phone service). Use the username and pin assigned to the voicemail box in Unity Express for Login (authentication). Once logged in: 1. Press 1 for Inbox 2. Highlight and Select desired message 3. Press Listen At that point the message should be played via the speakerphone. Please let us know your results. HTH, Amy --- Amy Ryan CCIE #24677 (Voice) Technical Instructor - IPexpert, Inc. Mailto: ar...@ipexpert.com Telephone: +1.810.326.1444 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat eFax: +1.810.454.0130 IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ From: Sergio Polizer spoli...@hotmail.com Date: Mon, 12 Apr 2010 18:10:36 -0300 To: moataz_m...@yahoo.com, mlin...@conet.de Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Vol 2 Lab 10 Q4.6 - VoiceView and CUCM - Authentication Error Hello Mirco, I added 4 IVR ports but same issue. Any other suggestions? Can It be an expected behavior? Thank you, Sergio. cue# sh software licenses Installed license files: - voicemail_lic.sig : 12 MAILBOX LICENSE - ivr_lic.sig : 4 PORT IVR BASE LICENSE Core: - Application mode: CCM - Total usable system ports: 8 Voicemail/Auto Attendant: - Max system mailbox capacity time: 6000 - Default # of general delivery mailboxes: 5 - Default # of personal mailboxes: 12 - Max # of configurable mailboxes: 17 Interactive Voice Response: - Max # of IVR sessions: 4 Languages: - Max installed languages: 5 - Max enabled languages: 5 Date: Mon, 12 Apr 2010 10:27:22 -0700
[OSL | CCIE_Voice] UCCX Skill Groups
Does anyone know if there is a way to define two skill groups for an agent, but give a particular skill group a priority over the other one? Meaning, if two calls are in queue, each for different skill groups, I can adjust which call will get priority over the other based on the type of skill? Thanks! Matthew Berry, CCVP, Sr. Unified Communications Engineer Kroll | 9023 Columbine Road, Eden Prairie, MN 55347 Single Number Reach +1 952 516 3748 | Fax +1 952 516 3646 | mjbe...@kroll.commailto:david.ra...@kroll.com www.krollontrack.comhttp://www.krollbackgroundscreening.com/ | www.kroll.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCCX Skill Groups
Here's what I am proposing: CSQ-ClientCare with a Resource Selection Criteria of Most Skilled by Weight I would then define the following skills: SKL-ClientCare0 Min comp 5 Weight 100 SKL-ClientCare1 Min comp 5 Weight 50 I would have two agents John Adams Assigned Skill= SKL-ClientCare0 Tom Jefferson Assigned Skill= SKL-ClientCare1 If a call came in and was sent to CSQ-ClientCare, the call should be routed to John Adams before Tom Jefferson since SKL-ClientCare0 had a higher weight that SKL-ClientCare1. Does that sound correct? Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.com -Original Message- From: bkvalent...@gmail.com [mailto:bkvalent...@gmail.com] Sent: Monday, April 19, 2010 9:21 AM To: Berry, Matthew J.; ccie_voice-boun...@onlinestudylist.com; OSL Subject: Re: [OSL | CCIE_Voice] UCCX Skill Groups Yes. You can weight the skills when you setup the CSQ. Brian Sent via BlackBerry from T-Mobile -Original Message- From: Berry, Matthew J. mjbe...@krollontrack.com Date: Mon, 19 Apr 2010 09:19:33 To: OSLccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] UCCX Skill Groups ___ 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] Meet Me
Wilson, You Meet-Me numbers will need to belong to a partition that the inbound gateway CSS can see. Make sure that the significant digits set on the gateway will deliver the correct number of digits to CUCM to match up with the Meet-Me ranges. To my knowledge, there is no way to set a PIN to a Meet-Me directly. Your best bet would be to restrict access via Partitions and CSS. Although, you could probably setup translation patterns in your PT-Internal partition to send calls to a PT-MeetMe partition with a FAC code enabled. The only caveat there is that you'd need to figure out a way for users to be able to initiate a Meet-Me conference with that configuration. Best of luck. Are you studying for your CCIE? Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.commailto:david.ra...@kroll.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Wilson Samuel Sent: Monday, April 19, 2010 3:49 PM To: OSL Subject: [OSL | CCIE_Voice] Meet Me Hi, I'm new to Meet Me, and was wondering once the Meet Me Numbering is set up, how do I ensure that callers from outside can participate in the conf. Also is there a possibility to set up a PIN or password for the conference (I presume yes) Regards Sam ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Call-Agent Time-of-Day Routing - CUCME and CUE
From the CUE Design Guide: Call-Agent Time-of-Day Routing: Time-of-Day (ToD) routing of calls to a receptionist (in contrast to the AA) requires a ToD routing feature on your call agent. With Cisco CME, this can be done by using a Tool Command Language (TCL) 2.0 script named Time of Day Routing and Barring that is available on Cisco.com Developer Support Central under TCL 2.0 technologies (this page requires a login). This feature is also available with Cisco CallManager Release 4.1. Question: Has anyone configure the Time of Day Routing and Barring TCL script? Any liklihood to see this on the exam? I am interested in anyone's feedback. Thanks! Matthew Berry, CCVP, Sr. Unified Communications Engineer Kroll | 9023 Columbine Road, Eden Prairie, MN 55347 Single Number Reach +1 952 516 3748 | Fax +1 952 516 3646 | mjbe...@kroll.commailto:david.ra...@kroll.com www.krollontrack.comhttp://www.krollbackgroundscreening.com/ | www.kroll.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] cme 7.0
You will need to add a voice translation pattern to append the + to all incoming/outgoing calls. Then, apply the rule to the voice-port for your T1/E1. Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.commailto:david.ra...@kroll.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of anupam TYAGI Sent: Friday, April 16, 2010 12:04 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] cme 7.0 Hi Does, cme version 7.0 support + sign. i can see in the isdn debug that the calling number is coming with + , but on the missed or received call it is with out +.. Is there any way to achieve it Thanks Anupam ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] [cisco-voip] Plz help, anyone knows the order of operation of Calling party transformation patterns in CUCM 7.0.1?
It took me two days to realize that EPNM = External Phone Number Mask #fail :) Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.commailto:david.ra...@kroll.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roger Källberg Sent: Thursday, April 15, 2010 2:02 AM To: jeremy co; le...@uoguelph.ca Cc: ccie_voice@onlinestudylist.com; cisco-v...@puck.nether.net Subject: Re: [OSL | CCIE_Voice] [cisco-voip] Plz help, anyone knows the order of operation of Calling party transformation patterns in CUCM 7.0.1? Hi Jeremy, You got this answer from Otto Sanchez on your other post on the same topic. Hello Jeremy, I think that an important fact to know is that cg/cd xform patterns get matched at the time the rp is being hit, so cg/cd xforms will override the transformations you perform at the rp level, If you configure the EPNM in a TP, the cg number getting to the rp will be the globalized number, right?, in this case, a cg xform pattern will be matched only if it corresponds to that EPNM, Please let me know if this clears up things a little bit for you, I would say that the answer you got is pretty strait forward, but if not test it out in all sort of ways in your lab and I think it will click for you. Best regards Roger Källberg Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ Från: jeremy co [jeremy.coo...@gmail.com] Skickat: den 15 april 2010 04:22 Till: le...@uoguelph.ca Kopia: ccie_voice@onlinestudylist.com; cisco-v...@puck.nether.net Ämne: Re: [OSL | CCIE_Voice] [cisco-voip] Plz help, anyone knows the order of operation of Calling party transformation patterns in CUCM 7.0.1? well, I know this order but CngPTP break these rules. Check out this scenario: callTPRP--- SLRG-GW ,check EPNM on TP and RP ,put CngPTP on GW.it overrides. ,check EPNM only on RP ,put CngPTP on GW.it does not override. ,check EPNM only on TP ,put CngPTP on GW.it overrides. How these mess works with CngPTP? Jeremy On Thu, Apr 15, 2010 at 12:13 PM, le...@uoguelph.camailto:le...@uoguelph.ca wrote: The farther down the chain you go the override happens. So: gateway overrides RL RL overrides RG RG overrides RP RP overrides phone Pretty sure about this, but I'm sure someone will chime in. Lelio Fulgenzi, Senior Analyst Computing Communications University of Guelph 519-824-4120 x56354 ...sent from my iPod - please pardon my fat fingers ;) [XKJ2000] On 2010-04-14, at 10:09 PM, jeremy co jeremy.coo...@gmail.commailto:jeremy.coo...@gmail.com wrote: Hi, Anyone knows how calling party transformation pattern order of operation in CUCM 7.0.1? I just got confused and docs are pretty weak in this area. It should override RP,RG,RL configuration. But apparently it would not! What is the relationship of checking EPNM on TPRP,GW and CngPTP ? I really appreciate any comment on this. Any doc that explain this relationship? Cheers Jeremy ___ cisco-voip mailing list cisco-v...@puck.nether.netmailto:cisco-v...@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] QSIG on the lab?
I don't know much about QSIG, so pardon the simple minded question: Could QSIG be on the lab and, if so, in what ways could we see it manifest? Thanks! - Sent from my Blackberry ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 1 Lab 8 - Connection drops after 50 seconds
I actually just emailed Ben. We'll see. - Sent from my Blackberry From: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com To: roger.kallb...@cygate.se roger.kallb...@cygate.se; ciscovoiceg...@gmail.com ciscovoiceg...@gmail.com; CCIE OSL ccie_voice@onlinestudylist.com Sent: Thu Apr 01 18:12:19 2010 Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 8 - Connection drops after 50 seconds I don't want to hi-jack this thread, but have to ask... Roger, have you heard anything regarding your HAT? ...and isn't Matthew supposed to get one too? :) -Marty From: roger.kallb...@cygate.se To: ciscovoiceg...@gmail.com; ccie_voice@onlinestudylist.com Date: Thu, 1 Apr 2010 16:48:19 +0200 Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 8 - Connection drops after 50 seconds Hi Matthew, You probably need this command no mgcp timer receive-rtcp, allthough it was mentioned during the IPX 10-day ILT held the last couple of weeks that it shouldn't be needed in the IOS version we are running in the lab. Brgds, Roger Källberg Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ Direkt: +46108787498 Växel: +46108787400 roger.kallb...@cygate.semailto:roger.kallb...@cygate.se Från: Matthew Berry [ciscovoiceg...@gmail.com] Skickat: den 1 april 2010 16:11 Till: OSL Ämne: [OSL | CCIE_Voice] Vol 1 Lab 8 - Connection drops after 50 seconds All - I cannot maintain a call between my BR1 Phone 1 and PSTN Phone over my MGCP gateway for more than 50 seconds. Is there a certain timer that I am running into? I do not encounter this issue through H.323 gateway. -- Matthew Berry A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Written Gmail: ciscovoiceguru Skype: ciscovoiceguru Twitter: ciscovoiceguru 1st Lab Attempt: Aug 16, 2010 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] SIP SRST - What application to use?
All - Here's a sample section from a SIP SRST setup from the SIP SRND Admin Guide: voice register pool 1 id network 10.10.201.0 mask 255.255.255.0 application sip.app preference 2 incoming called-number cor incoming css-internal default codec g711ulaw What the heck is this application command used for? Later on, I came across this config example: ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP SRST - What application to use?
So what you're saying is that SIP SRST seems to work properly even without the sip.app application specified? I haven't been able to tell a different without the application, which is what raised the question about its function. M From: Angel Perez [mailto:gorr...@hotmail.com] Sent: Thursday, March 25, 2010 6:16 AM To: Berry, Matthew J.; osl osl Subject: RE: [OSL | CCIE_Voice] SIP SRST - What application to use? Hello: The second example is not shown... My experience tell me that if you use application sip.app the gw won't find the app, then you will need application global service alternate Default (similar to mgcp srst) this way the gw will use h323 and call will work. A better aproach that worked for me is just delete this command application sip.app I know that this doesn't answer your question but could help Regards From: mjbe...@krollontrack.com To: ccie_voice@onlinestudylist.com Date: Thu, 25 Mar 2010 06:00:35 -0500 Subject: [OSL | CCIE_Voice] SIP SRST - What application to use? All - Here's a sample section from a SIP SRST setup from the SIP SRND Admin Guide: voice register pool 1 id network 10.10.201.0 mask 255.255.255.0 application sip.app preference 2 incoming called-number cor incoming css-internal default codec g711ulaw What the heck is this application command used for? Later on, I came across this config example: Hotmail: Trusted email with Microsoft's 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
Re: [OSL | CCIE_Voice] SRST display problem
Ashar - You need to change the display name on your ephone. The +617 Is in the place where you'd normally put the caller ID. Your are displaying the digits correctly. You just need to change the CID. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ashar Sent: Sunday, March 21, 2010 10:54 AM To: CCIE Voice OSL (ccie_voice@onlinestudylist.com) Subject: [OSL | CCIE_Voice] SRST display problem Hello, I am having some issues in displaying the number on PSTN phone while in SRST mode. I only want the calling number to get displayed on PSTN phone but I am getting a number with + and then number again in brackets for e.g. it display this on PSTN phone From +16178631001 (6178631001) while I want it to display like From 6178631001. Please see the debug: Mar 21 19:34:04.775: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type 0xD is 0x0 0x1, Calling num 6178631001 Mar 21 19:34:04.775: ISDN Se0/0/0:23 Q931: Sending SETUP callref = 0x0094 callID = 0x8015 switch = primary-ni interface = User Mar 21 19:34:04.779: ISDN Se0/0/0:23 Q931: TX - SETUP pd = 8 callref = 0x0094 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Progress Ind i = 0x8183 - Origination address is non-ISDN Display i = '+16178631001' Calling Party Number i = 0x0180, '6178631001' Plan:ISDN, Type:Unknown Called Party Number i = 0xA1, '12123945001' Plan:ISDN, Type:National Mar 21 19:34:04.811: ISDN Se0/0/0:23 Q931: RX - CALL_PROC pd = 8 callref = 0x8094 Channel ID i = 0xA98381 Exclusive, Channel 1 Mar 21 19:34:04.871: ISDN Se0/0/0:23 Q931: RX - ALERTING pd = 8 callref = 0x8094 Progress Ind i = 0x8088 - In-band info or appropriate now available --More-- Mar 21 19:34:08.819: ISDN Se0/0/0:23 Q931: TX - DISCONNECT pd = 8 callref = 0x0094 Cause i = 0x8090 - Normal call clearing Mar 21 19:34:08.831: ISDN Se0/0/0:23 Q931: RX - RELEASE pd = 8 callref = 0x8094 Mar 21 19:34:08.835: ISDN Se0/0/0:23 Q931: TX - RELEASE_COMP pd = 8 callref = 0x0094 I am not sure but that Display bit in red above might be the problem. I am using following telephony-service commands: telephony-service srst mode auto-provision none srst dn line-mode dual max-ephones 5 max-dn 5 ip source-address 10.10.201.1 port 2000 max-redirect 10 voicemail 912123945220 max-conferences 2 gain -6 moh music-on-hold.au transfer-system full-consult transfer-pattern .T secondary-dialtone 9 create cnf-files Is there a way to get rid of that display thing? Does it depend on PSTN configuration? Any help would be much appreciated. -- Thanks, Ashar Siddiqui ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP Phone registration problem - SIP/2.0 401 Unauthorized message
Do you have the authenticate register command entered? - Sent from my Blackberry From: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Sat Mar 20 09:30:11 2010 Subject: [OSL | CCIE_Voice] SIP Phone registration problem - SIP/2.0 401 Unauthorized message Hi guys, I am having a hard time registering my home SIP phone with CUCME on BR2 RTR. Weird thing is the phone display has the DN number, I get a dial tone and I can dial from the phone to the BR2 SCCP Phone. However, I can't receive any phone calls and the output of the show voice register pool 2 is Output of deb ccsip message REGISTER sip:10.10.202.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.12.13:5060;branch=z9hG4bK58ea2724 From: sip:3...@10.10.202.1mailto:sip%3a3...@10.10.202.1;tag=003094c2f26b663e5919-699325d0 To: sip:3...@10.10.202.1mailto:sip%3a3...@10.10.202.1 Call-ID: 003094c2-f26b-0a633263-5f6ad...@192.168.12.13mailto:003094c2-f26b-0a633263-5f6ad...@192.168.12.13 Max-Forwards: 70 CSeq: 205 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: sip:3...@192.168.12.13:5060;user=phone;transport=udp;+sip.instance=urn:uuid:----003094c2f BR2-RTR#200;+u.sip!model.ccm.cisco.comhttp://model.ccm.cisco.com=7 Supported: replaces,join,norefersub,X-cisco-callinfo,X-cisco-service-control Content-Length: 0 Expires: 3600 Mar 20 14:09:45.663: //-1//SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.12.13:5060;branch=z9hG4bK58ea2724 From: sip:3...@10.10.202.1mailto:sip%3a3...@10.10.202.1;tag=003094c2f26b663e5919-699325d0 To: sip:3...@10.10.202.1mailto:sip%3a3...@10.10.202.1 Date: Sat, 20 Mar 2010 14:09:45 GMT Call-ID: 003094c2-f26b-0a633263-5f6ad...@192.168.12.13mailto:003094c2-f26b-0a633263-5f6ad...@192.168.12.13 Server: Cisco-SIPGateway/IOS-12.x CSeq: 205 REGISTER Content-Length: 0 Mar 20 14:09:46.163: //-1//SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.12.13:5060;branch=z9hG4bK58ea2724 From: sip:3...@10.10.202.1mailto:sip%3a3...@10.10.202.1;tag=003094c2f26b663e5919-699325d0 To: sip:3...@10.10.202.1mailto:sip%3a3...@10.10.202.1;tag=7DED54-1B43 Date: Sat, 20 Mar 2010 14:09:45 GMT Call-ID: 003094c2-f26b-0a633263-5f6ad...@192.168.12.13mailto:003094c2-f26b-0a633263-5f6ad...@192.168.12.13 Server: Cisco-SIPGateway/IOS-12.x CSeq: 205 REGISTER WWW-Authenticate: Digest realm=,nonce=94FB834E000C97EE,algorithm=MD5,qop=auth Content-Length: 0 Mar 20 14:09:46.259: //-1//SIP/Msg/ccsipDisplayMsg: Received: REGISTER sip:10.10.202.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.12.13:5060;branch=z9hG4bK58ea2724 From: sip:3...@10.10.202.1mailto:sip%3a3...@10.10.202.1;tag=003094c2f26b663e5919-699325d0 To: sip:3...@10.10.202.1mailto:sip%3a3...@10.10.202.1 Call-ID: 003094c2-f26 br2-rtr#b-0a633263-5f6ad...@192.168.12.13mailto:b-0a633263-5f6ad...@192.168.12.13 Max-Forwards: 70 CSeq: 205 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: sip:3...@192.168.12.13:5060;user=phone;transport=udp;+sip.instance=urn:uuid:----003094c2f200;+u.sip!model.ccm.cisco.comhttp://model.ccm.cisco.com=7 Supported: replaces,join,norefersub,X-cisco-callinfo,X-cisco-service-control Content-Length: 0 Expires: 3600 BR2-RTR#show voice register pool 2 Pool Tag 2 Config: Mac address is 0030.94C2.F200 Type is 7960 Number list 1 : DN 2 Proxy Ip address is 0.0.0.0 DTMF Relay is enabled, rtp-nte Call Waiting is enabled DnD is disabled Description is 32143006 keep-conference is enabled username cisco password 123 template is 1 service-control mechanism is not supported Dialpeers created: Statistics: Active registrations : 0 Total SIP phones registered: 0 Total Registration Statistics Registration requests : 0 Registration success : 0 Registration failed : 0 unRegister requests : 0 unRegister success : 0 unRegister failed : 0 SHOW RUN Config from the BR2 RTR is voice register global mode cme source-address 10.10.202.1 port 5060 max-dn 2 max-pool 2 load 7960-7940 P0S3-08-6-00 timezone 13 time-format 24 date-format D/M/Y voicemail 3600 tftp-path flash: create profile sync 0001011653248495 ntp-server 10.10.100.2 mode unicast ! voice register dn 1 number 3005 name br2 phn 3 ! voice register dn 2 number 3006 name br2 phn 4 ! voice register template 1 dialplan 1 no conference enable ! voice register dialplan 1 type 7940-7960-others pattern 1 3... pattern 2 999 ! voice register pool 1 id mac 0011.BBEF.6FB9 type 7960 number 1 dn 1 template 1 dtmf-relay rtp-nte username 3005 password cisco description 32143005 codec g711ulaw ! voice register pool 2 id mac 0030.94C2.F200 type 7960 number 1 dn 2 template 1 username cisco password 123 description 32143006 codec g711ulaw I also removed the username and password and tried that. Still the same result. I have tried logging into the phone and added the authentication name
Re: [OSL | CCIE_Voice] SIP Phone registration problem - SIP/2.0 401 Unauthorized message
Whoops. Should have read Otto's email. Sorry for the duplicate. - Sent from my Blackberry From: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com To: Kalyan iyer kparam2...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Sat Mar 20 10:47:16 2010 Subject: Re: [OSL | CCIE_Voice] SIP Phone registration problem - SIP/2.0 401 Unauthorized message Hi Kalyan, authenticate register command is missing from you voice register global configuration, also, be sure that the registrar server and bind commands under sip configuration mode are properly implemented, If still with issues, please send full configuration and sip debugs, Thanks, On Sat, Mar 20, 2010 at 10:00 AM, Kalyan iyer kparam2...@gmail.commailto:kparam2...@gmail.com wrote: Hi guys, I am having a hard time registering my home SIP phone with CUCME on BR2 RTR. Weird thing is the phone display has the DN number, I get a dial tone and I can dial from the phone to the BR2 SCCP Phone. However, I can't receive any phone calls and the output of the show voice register pool 2 is Output of deb ccsip message REGISTER sip:10.10.202.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.12.13:5060;branch=z9hG4bK58ea2724 From: sip:3...@10.10.202.1mailto:sip%3a3...@10.10.202.1;tag=003094c2f26b663e5919-699325d0 To: sip:3...@10.10.202.1mailto:sip%3a3...@10.10.202.1 Call-ID: 003094c2-f26b-0a633263-5f6ad...@192.168.12.13mailto:003094c2-f26b-0a633263-5f6ad...@192.168.12.13 Max-Forwards: 70 CSeq: 205 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: sip:3...@192.168.12.13:5060;user=phone;transport=udp;+sip.instance=urn:uuid:----003094c2f BR2-RTR#200;+u.sip!model.ccm.cisco.comhttp://model.ccm.cisco.com=7 Supported: replaces,join,norefersub,X-cisco-callinfo,X-cisco-service-control Content-Length: 0 Expires: 3600 Mar 20 14:09:45.663: //-1//SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.12.13:5060;branch=z9hG4bK58ea2724 From: sip:3...@10.10.202.1mailto:sip%3a3...@10.10.202.1;tag=003094c2f26b663e5919-699325d0 To: sip:3...@10.10.202.1mailto:sip%3a3...@10.10.202.1 Date: Sat, 20 Mar 2010 14:09:45 GMT Call-ID: 003094c2-f26b-0a633263-5f6ad...@192.168.12.13mailto:003094c2-f26b-0a633263-5f6ad...@192.168.12.13 Server: Cisco-SIPGateway/IOS-12.x CSeq: 205 REGISTER Content-Length: 0 Mar 20 14:09:46.163: //-1//SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.12.13:5060;branch=z9hG4bK58ea2724 From: sip:3...@10.10.202.1mailto:sip%3a3...@10.10.202.1;tag=003094c2f26b663e5919-699325d0 To: sip:3...@10.10.202.1mailto:sip%3a3...@10.10.202.1;tag=7DED54-1B43 Date: Sat, 20 Mar 2010 14:09:45 GMT Call-ID: 003094c2-f26b-0a633263-5f6ad...@192.168.12.13mailto:003094c2-f26b-0a633263-5f6ad...@192.168.12.13 Server: Cisco-SIPGateway/IOS-12.x CSeq: 205 REGISTER WWW-Authenticate: Digest realm=,nonce=94FB834E000C97EE,algorithm=MD5,qop=auth Content-Length: 0 Mar 20 14:09:46.259: //-1//SIP/Msg/ccsipDisplayMsg: Received: REGISTER sip:10.10.202.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.12.13:5060;branch=z9hG4bK58ea2724 From: sip:3...@10.10.202.1mailto:sip%3a3...@10.10.202.1;tag=003094c2f26b663e5919-699325d0 To: sip:3...@10.10.202.1mailto:sip%3a3...@10.10.202.1 Call-ID: 003094c2-f26 br2-rtr#b-0a633263-5f6ad...@192.168.12.13mailto:b-0a633263-5f6ad...@192.168.12.13 Max-Forwards: 70 CSeq: 205 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: sip:3...@192.168.12.13:5060;user=phone;transport=udp;+sip.instance=urn:uuid:----003094c2f200;+u.sip!model.ccm.cisco.comhttp://model.ccm.cisco.com=7 Supported: replaces,join,norefersub,X-cisco-callinfo,X-cisco-service-control Content-Length: 0 Expires: 3600 BR2-RTR#show voice register pool 2 Pool Tag 2 Config: Mac address is 0030.94C2.F200 Type is 7960 Number list 1 : DN 2 Proxy Ip address is 0.0.0.0 DTMF Relay is enabled, rtp-nte Call Waiting is enabled DnD is disabled Description is 32143006 keep-conference is enabled username cisco password 123 template is 1 service-control mechanism is not supported Dialpeers created: Statistics: Active registrations : 0 Total SIP phones registered: 0 Total Registration Statistics Registration requests : 0 Registration success : 0 Registration failed : 0 unRegister requests : 0 unRegister success : 0 unRegister failed : 0 SHOW RUN Config from the BR2 RTR is voice register global mode cme source-address 10.10.202.1 port 5060 max-dn 2 max-pool 2 load 7960-7940 P0S3-08-6-00 timezone 13 time-format 24 date-format D/M/Y voicemail 3600 tftp-path flash: create profile sync 0001011653248495 ntp-server 10.10.100.2 mode unicast ! voice register dn 1 number 3005 name br2 phn 3 ! voice register dn 2 number 3006 name br2 phn 4 ! voice register template 1 dialplan 1 no conference enable ! voice register dialplan 1 type 7940-7960-others pattern 1 3... pattern 2 999 ! voice register
Re: [OSL | CCIE_Voice] Translation Pattern: Negate a range
Angel, For starters, the only scenario where you're need to worry about a plus being sent to an IOS gateway from CUCM would be an MGCP/SIP gateway. H.323 cannot receive a plus from CUCM; it will simply strip it off before it hits any translation rules. That said, say you are asked to strip off a + on a SIP/H.323 based CUCME router for all inbound calls. If you're running a negation in a translation pattern, you should enter the command like this: voice translation-rule 1 rule 1 /\+\([^1^2].*\)/ /\1/ Let's break this apart: rule 1 / \+ \( [^1^2].* \) / / \1 / * Rule 1 * Start match pattern * \+ removes the special significance from the + and treats it as a digit * Now we need to create match pattern section. We use parenthesis prefixed by \ to remove special significance * Your negation pattern requires a caret (^) for every negated value. [^123] would negate 1 but allow 2 and 3. [^1^2] negates both 1 and 2. * .* functions just like the ! in CUCM * Now we end the match pattern section. We use parenthesis prefixed by \ to remove special significance * Start replace pattern * \1 for match pattern scenario #1 test voice translation-rule 1 +2000 +2000 Didn't match with any of rules test voice translation-rule 1 +1000 +1000 Didn't match with any of rules test voice translation-rule 1 +4000 Matched with rule 1 Original number: +4000 Translated number: 4000 Original number type: none Translated number type: none Original number plan: none Translated number plan: none test voice translation-rule 1 +400043324324 Matched with rule 1 Original number: +400043324324 Translated number: 400043324324 Original number type: none Translated number type: none Original number plan: none Translated number plan: none Does that help? From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Angel Perez Sent: Friday, March 19, 2010 6:30 AM To: osl osl Subject: [OSL | CCIE_Voice] Translation Pattern: Negate a range Hi: I wan't to negate a range of numbers in a tranlation pattern like in a router pattern, but I think that this is not possible after doing some tests. For example, I want to match all pattern that begin with + and then any number except 1 and 2 \+[^1-2]! In a router pattern this would match, but in a translation pattern do not... Any suggestions? Thanks 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] Need Clarification: mls qos map cos-dscp
The QoS SRND states that the auto qos voip command adds the following config to the router: C2970(config)# mls qos C2970(config)# mls qos map cos-dscp 0 8 16 26 32 46 48 56 Earlier in the SRND, around page 40, it says that the old marking for audio signaling was AF31 (26). That is the same DSCP marking listed above. As part of our best-practice scenario, should we be changing the command to consider audio signaling as CS3 (24)? The command would need to be modified: C2970(config)# mls qos map cos-dscp 0 8 16 24 32 46 48 56 Is this true? Otto, can you weigh in on this one? Thanks! Matthew Berry Digital Footprint: Twitter: ciscovoiceguru Skype: ciscovoiceguru 1st Lab Attempt: Aug 16th, 2010 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] QoS SRND - Page 105
Pulling from the QoS SRND, the following configuration is only supposed to allow the bandwidth for one voice call per switchport VLAN. Obviously, based on the 128k, we're focused on G.711 calls (so my next question will not apply to G.729). I want to know if the following command would disable the ability to have multiple calls (different lines) on the same phone. For example: Phone A (with the policing command below) calls Phone B. At this point, 128k of G.711 bandwidth is consumed. If Phone A puts Phone B on hold and calls Phone C, would the call no go through due to policing? CAT2970(config-cmap)#policy-map IPPHONE+PC-BASIC CAT2970(config-pmap)#class VVLAN-VOICE CAT2970(config-pmap-c)# set ip dscp 46 ! DSCP EF (Voice) CAT2970(config-pmap-c)# police 128000 8000 exceed-action drop I guess what I am asking is what happens to the initial call when it is placed on hold? Is the audio stream maintained between phones (128k), thereby eliminating the ability to place another call? Matthew Berry Digital Footprint: Twitter: ciscovoiceguru Skype: ciscovoiceguru 1st Lab Attempt: Aug 16th, 2010 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] mwi in srst
Anupam, Can you please provide more detail? Take opportunities like this to thoroughly explain what you do know, including config examples and outputs from show statements. Your issues could be caused by any number of nuances. Approach these scenarios as if you were already a CCIE. Provide detail. No one can help you unless you do your due diligence in sending us usable information. It's a lot like opening a case with Cisco TAC. If you tell them, My Cisco is broke, they're going to throw your question on the back burner and send you canned email responses. If you give them usable information, they'll go the extra mile because you have already learned their respect as someone who knows what's going on. Matthew Berry From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of anupam TYAGI Sent: Thursday, March 18, 2010 12:29 PM To: ccie_voice@onlinestudylist.com; ccie_voice-requ...@onlinestudylist.com Subject: [OSL | CCIE_Voice] mwi in srst Hi Guys, My MWI is not working in SRST . what can be the probable reasons .. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] about globalization and the lab's PSTN
Jean, Are you using Proctor Labs or your own lab? The PSTN router should take care of the calling number type. You should also make sure you don't have any translation patterns on the BR2 gateway that would modify the type. Also check your H323 gateway to ensure the same thing. - Sent from my Blackberry From: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com To: 'ccie_voice@onlinestudylist.com' ccie_voice@onlinestudylist.com Sent: Wed Mar 17 19:21:33 2010 Subject: [OSL | CCIE_Voice] about globalization and the lab's PSTN Hello All. I am experiencing the following behavior: I place a call out of the Brach 2 site, internationally into the HQ site, the PSTN sends the call into HQ as “national”. If I place the call from the PSTN phone international line (India or Spain) into the HQ site, the call comes in correctly labeled as “international”. Is this the expected behavior due to the simulation of the PSTN in the lab? Or I am not setting something that I should when routing the calls out the Branch2 site? Any advice is greatly appreciated. Regards! MT ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] about globalization and the lab's PSTN
That's correct. When the call is delivered to the HQ gateway it is seen as national from the perspective of the terminating ISDN. From the standpoint of HQ, the 212 is national type. If it was a local PRI from a local LEC, I might expect a subscriber type, but itd be most common to receive a national type. - Sent from my Blackberry From: Jean M. Thewissen m...@mnet.com.mx To: Berry, Matthew J.; 'ccie_voice@onlinestudylist.com' ccie_voice@onlinestudylist.com Sent: Wed Mar 17 20:55:13 2010 Subject: RE: [OSL | CCIE_Voice] about globalization and the lab's PSTN I am using proctorlabs… What really confuses me is that when the call leaves the B2 GW, it is correctly tagged as international. Bearer Capability i = 0x8090A3 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98383 Exclusive, Channel 3 Display i = 'b2 phone 3' Calling Party Number i = 0x0081, '+3432143003' Plan:Unknown, Type:Unknown Called Party Number i = 0x91, '0012123945001' Plan:ISDN, Type:International But when PSTN sends it to HQ GW, it is tagged as national: Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Display i = 'b2 phone 3' Calling Party Number i = 0x0081, '+3432143003' Plan:Unknown, Type:Unknown Called Party Number i = 0xA1, '2123945001' Plan:ISDN, Type:National I really don’t see how I could alter how the PSTN tags the call… but maybe I am just not seeing the whole picture. From: Berry, Matthew J. [mailto:mjbe...@krollontrack.com] Sent: miércoles, 17 de marzo de 2010 07:50 p.m. To: Jean M. Thewissen; 'ccie_voice@onlinestudylist.com' Subject: Re: [OSL | CCIE_Voice] about globalization and the lab's PSTN Jean, Are you using Proctor Labs or your own lab? The PSTN router should take care of the calling number type. You should also make sure you don't have any translation patterns on the BR2 gateway that would modify the type. Also check your H323 gateway to ensure the same thing. - Sent from my Blackberry From: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com To: 'ccie_voice@onlinestudylist.com' ccie_voice@onlinestudylist.com Sent: Wed Mar 17 19:21:33 2010 Subject: [OSL | CCIE_Voice] about globalization and the lab's PSTN Hello All. I am experiencing the following behavior: I place a call out of the Brach 2 site, internationally into the HQ site, the PSTN sends the call into HQ as “national”. If I place the call from the PSTN phone international line (India or Spain) into the HQ site, the call comes in correctly labeled as “international”. Is this the expected behavior due to the simulation of the PSTN in the lab? Or I am not setting something that I should when routing the calls out the Branch2 site? Any advice is greatly appreciated. Regards! MT ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MVA Disconnects after prompt
How many seconds go by? Could you be hitting the T38 timeout value? Perhaps something with call capabilities are not being setup? - Sent from my Blackberry - Original Message - From: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com To: Jason Granat j...@slash128.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Wed Mar 17 21:12:44 2010 Subject: Re: [OSL | CCIE_Voice] MVA Disconnects after prompt On a side note SNR works to the same remote destination from the same phone, so I would imagine my partitions and CSS's are correct... Sent while mobile. On Mar 17, 2010, at 19:09, Jason Granat j...@slash128.com wrote: I've seen this discussed quite a bit. I have had it working, but after another attempt from scratch I am able to dial the DID from a remote destination, I get the 'welcome' message, then after a few seconds the call disconnects. If I call from a non-remote destination I get the prompt to input my remote destination and then after a few seconds it drops. I've debugged the script, ISDN, dial-peers, etc., but having a mental block. Is this a symptom of a common problem that I am drawing a blank on? Sent while mobile. http://slash128.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://slash128.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] MVA Disconnects after prompt
Sorry. Thinking of wrong term. Sounds like capabilities are not being established. Are you using h323 fast start? - Sent from my Blackberry - Original Message - From: Jason Granat j...@slash128.com To: Berry, Matthew J. Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Wed Mar 17 21:32:12 2010 Subject: Re: [OSL | CCIE_Voice] MVA Disconnects after prompt 10 seconds on the nose every time. Isn't T.38 for fax? Sent while mobile. On Mar 17, 2010, at 19:20, Berry, Matthew J. mjbe...@krollontrack.com wrote: How many seconds go by? Could you be hitting the T38 timeout value? Perhaps something with call capabilities are not being setup? - Sent from my Blackberry - Original Message - From: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com To: Jason Granat j...@slash128.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Wed Mar 17 21:12:44 2010 Subject: Re: [OSL | CCIE_Voice] MVA Disconnects after prompt On a side note SNR works to the same remote destination from the same phone, so I would imagine my partitions and CSS's are correct... Sent while mobile. On Mar 17, 2010, at 19:09, Jason Granat j...@slash128.com wrote: I've seen this discussed quite a bit. I have had it working, but after another attempt from scratch I am able to dial the DID from a remote destination, I get the 'welcome' message, then after a few seconds the call disconnects. If I call from a non-remote destination I get the prompt to input my remote destination and then after a few seconds it drops. I've debugged the script, ISDN, dial-peers, etc., but having a mental block. Is this a symptom of a common problem that I am drawing a blank on? Sent while mobile. http://slash128.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://slash128.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://slash128.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] SRND Documentation for Unity Connection
Is anyone aware of good design documentation (akin to SRNDs) for Unity Connection? All I've been able to find is the following: * CUC Admin * CUC CUCM SCCP Integration Guide * CUC CUCME SCCP Integration Guide * CUC Design Guide (very limited, not very insightful) ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] QoS Calculation Value for L2 MLPoFR
Working on Vol 1 Lab 10A, Question 10.4 The Proctor Guide calculates L2 MLPoFR as 9 bytes per packet. However, the QoS SRND defines the following on page 1-15: - PPP = 12 bytes - MLP = 13 bytes - FR = 4 bytes - FR with FRF.12 = 8 bytes None of those match up. Why did IPexpert chose 9 bytes per packet? Matthew Berry Digital Footprint: Twitter: ciscovoiceguru Skype: ciscovoiceguru 1st Lab Attempt: Aug 16th, 2010 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] PSTN phone dialing in Vol 2 Lab 5
Might be simple, but do you have your H323 dial peers setup correctly? - Sent from my Blackberry From: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com To: Mike Todd michaelt...@gmail.com Cc: ccie_voice ccie_voice@onlinestudylist.com Sent: Sun Mar 14 20:57:51 2010 Subject: Re: [OSL | CCIE_Voice] PSTN phone dialing in Vol 2 Lab 5 Did you ever get this figured out? On Wed, Jan 6, 2010 at 1:14 PM, Mike Todd michaelt...@gmail.commailto:michaelt...@gmail.com wrote: I'm having problems figuring out how I'm supposed to be able to dial from certain lines to certain sites from the PSTN phone in this lab. I can dial fine from line 2 to the HQ site using 10 digit dialing, but when I try dialing from Line 3 or 4 to the same site I can't get any calls into the HQ router (no matter the way I dial). I've tried using the full E164 with and without various access codes (00, 000, 011, 900, 9000, 9011) and I get a busy signal for each call with no output on my HQ router debug ISDN q931. Any ideas? I'm sure I'm missing something stupid here... Thanks in advance! Mike Todd CCIE #10858 (Routing and Switching, Security) (and hopefully voice soon!) ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com -- www.ccietalk.comhttp://www.ccietalk.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Globalized + Dialing - Applying Called Party Number Type for Secondary Route Scenario (non-TEHO)
All - I am setting up + dialing on a self-made lab. A question has come up as to where the Called Party Number Type should be set. For this exercise, I want to find the best way to route calls through a system, utilizing alternate paths for failover scenarios. Those this does not take TEHO into account, I want a format that can easily accommodate TEHO situations. I believe my method below will do that. PSTN is expecting: Subscriber: Seven digits, Subscriber National: Eleven digits (incl. 1), National Intl: Undefined digits, Intl I have also setup translation patterns in PT_US_MN_EP_PSTN setup as: 9.952[2-9]XX-- Predot, Prefix +1 -- Result: +19525163748 (local MN) 9.1[2-9]XX[2-9]XX -- Predot, Prefix + -- Result: +1615444 (remote long-distance in TN) 9.011!-- Predot, Prefix +-- Result: +3432141861 (remote international) I have route patterns in PT_US_MN_EP_PSTN setup as: \+1952[2-9]XX-- +19525163748 \+1[2-9]XX[2-9]XX-- +1615444 \+! -- +3432141861 \+!# -- +3432141861 Note: I am not using predot in the route patterns!! At this point, all dialed numbers have been globalized from their localized variants. I am trying to figure out if it would be best to apply the Called Party Number Type (Sub, Nat, Intl) at the Translation Pattern, Route Pattern, Route List (i.e. Route Group settings), or Called Party Transformation on the gateway level. If I did not modify the Called Party Number Type at the Translation Pattern or Route Pattern, it would allow me to configure different call treatment at the route list level for TEHO. However, I think it would be best to apply that setting at the Called Party Transformation pattern on the MGCP gateways (avoiding H.323 for sake of conversation). I have two locations that calls will be sent out, MN and TN. For example: RL_US_MN_PSTN RG_US_MN RG_US_TN RL_US_TN_PSTN RG_US_TN RG_US_MN At this point, I need to convert the globalized numbers to their localized variants at the specific locations' voice gateways Called Party Transformation on MN gateway: +\1952.XXX -- Strip predot, Subscriber +\.1[2-9]XX[2-9]XX -- Strip predot, National +\.!-- Strip predot, Prefix 011, International Called Party Transformation on TN gateway: +\1615.XXX -- Strip predot, Subscriber +\.1[2-9]XX[2-9]XX -- Strip predot, National +\.!-- Strip predot, Prefix 011, International Primary Route (MN call out MN gateway) Verification: 1. User in MN dials local number as 9.9525163748 (my desk phone, give me a call :)) 2. Translation pattern changes to +19525163748 3. Matches route pattern of +1952[2-9]XX 4. Route pattern sent via RL_US_MN_PSTN to RG_US_MN 5. RG_US_MN sends call to US_MN_Gateway1 6. US_MN_Gateway1 has a called transformation pattern of +\1952.XXX (Subscriber) 7. Call goes out the US_MN_Gateway1 as 5163748 (Subscriber) Secondary Route (MN call out TN gateway) Verification: 1. User in MN dials local number as 9.9525163748 (my desk phone, give me a call :)) 2. Translation pattern changes to +19525163748 3. Matches route pattern of +1952[2-9]XX 4. Route pattern sent via RL_US_MN_PSTN to RG_US_TN (RG_US_MN is down and not functioning) 5. RG_US_TN sends call to US_TN_Gateway1 6. US_TN_Gateway1 has a called transformation pattern of +\.1[2-9]XX[2-9]XX (National) 7. Call goes out the US_TN_Gateway1 as 19525163748 (National) Any feedback would be appreciated. It took me about 30 minutes to think this through and type it out. Because it takes so long, I am trying to build a strawman structure that I can easily drop into the lab and modify to support my needs. What say ye? Matthew Berry ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MVA
Check the CSS on the remote destination profile you're calling from. If you do a debug isdn q931 on the PSTN gateway, do you see the call hit the gateway? Your rerouting CSS on the RDP is used for calls out to your RD. Your CSS on the RDP is used for calls through MVA that are routed out through your PSTN gateway. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of anupam TYAGI Sent: Friday, March 12, 2010 9:27 AM To: ccie_voice-requ...@onlinestudylist.com; ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] MVA Hi Folks I am doing MVA , When i dial the MVA number , I am able to hear the prompt. I dial a PSTN number , but the call disconnect . Can any body suggest me what can be the reason . Rgds Anu. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] UCCX/IVR Script Repository
Does anyone know of where Cisco's UCCX/IVR sample script repository is? I can't find it. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] calls from hq to and from cme sip phones
Omotayo, How do you have your regions setup in CUCM? The CUCME trunk through the HQ gateway should be placed in the HQ region. Can you also send me the HQ config as an attached file. Make sure your dspfarm has a 'no shutdown issued. Also, make sure your transcoder is registered to CUCM under Media Resources Transcoder. Did you also make sure the transcoder is configured as an IOS Enhanced Media Termination Point? Also, make sure you ALWAYS reset the trunk in CUCM. That will oftentimes clear out weird issues. I have learned that lesson the hard way. Matthew Berry From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Omotayo Sent: Wednesday, March 10, 2010 5:20 PM To: OSL Group Subject: Re: [OSL | CCIE_Voice] calls from hq to and from cme sip phones Hello, the debug shows that the codec getting to the cme is g711 because the codec byte is 160 and still it disconnects On Thu, Mar 11, 2010 at 12:03 AM, Omotayo adefilabi...@gmail.commailto:adefilabi...@gmail.com wrote: Hello, When i call from hq sccp phone to cme sip phone, it rings but when i pick up. it disconnects also when i call from cme sip phone to hq (sccp and sip) phone it rings on the hq phones when i pick t disconnects and contnues ringing on the sip phone i have a transcoder configured on the trunk Any one with a fix thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] calls from hq to and from cme sip phones
Have you issued a no shut' on dspfarm profile 1 transcode? From: Omotayo [mailto:adefilabi...@gmail.com] Sent: Wednesday, March 10, 2010 5:38 PM To: Berry, Matthew J. Cc: OSL Group Subject: Re: [OSL | CCIE_Voice] calls from hq to and from cme sip phones The trunk DP has a region that speaks g729 to hq and br1 find attached the config On Thu, Mar 11, 2010 at 12:24 AM, Berry, Matthew J. mjbe...@krollontrack.commailto:mjbe...@krollontrack.com wrote: Omotayo, How do you have your regions setup in CUCM? The CUCME trunk through the HQ gateway should be placed in the HQ region. Can you also send me the HQ config as an attached file. Make sure your dspfarm has a 'no shutdown issued. Also, make sure your transcoder is registered to CUCM under Media Resources Transcoder. Did you also make sure the transcoder is configured as an IOS Enhanced Media Termination Point? Also, make sure you ALWAYS reset the trunk in CUCM. That will oftentimes clear out weird issues. I have learned that lesson the hard way. Matthew Berry From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Omotayo Sent: Wednesday, March 10, 2010 5:20 PM To: OSL Group Subject: Re: [OSL | CCIE_Voice] calls from hq to and from cme sip phones Hello, the debug shows that the codec getting to the cme is g711 because the codec byte is 160 and still it disconnects On Thu, Mar 11, 2010 at 12:03 AM, Omotayo adefilabi...@gmail.commailto:adefilabi...@gmail.com wrote: Hello, When i call from hq sccp phone to cme sip phone, it rings but when i pick up. it disconnects also when i call from cme sip phone to hq (sccp and sip) phone it rings on the hq phones when i pick t disconnects and contnues ringing on the sip phone i have a transcoder configured on the trunk Any one with a fix thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] calls from hq to and from cme sip phones
If all else fails, save your configs and reboot BR2 and HQ routers. And test again. From: Omotayo [mailto:adefilabi...@gmail.com] Sent: Wednesday, March 10, 2010 5:38 PM To: Berry, Matthew J. Cc: OSL Group Subject: Re: [OSL | CCIE_Voice] calls from hq to and from cme sip phones The trunk DP has a region that speaks g729 to hq and br1 find attached the config On Thu, Mar 11, 2010 at 12:24 AM, Berry, Matthew J. mjbe...@krollontrack.commailto:mjbe...@krollontrack.com wrote: Omotayo, How do you have your regions setup in CUCM? The CUCME trunk through the HQ gateway should be placed in the HQ region. Can you also send me the HQ config as an attached file. Make sure your dspfarm has a 'no shutdown issued. Also, make sure your transcoder is registered to CUCM under Media Resources Transcoder. Did you also make sure the transcoder is configured as an IOS Enhanced Media Termination Point? Also, make sure you ALWAYS reset the trunk in CUCM. That will oftentimes clear out weird issues. I have learned that lesson the hard way. Matthew Berry From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Omotayo Sent: Wednesday, March 10, 2010 5:20 PM To: OSL Group Subject: Re: [OSL | CCIE_Voice] calls from hq to and from cme sip phones Hello, the debug shows that the codec getting to the cme is g711 because the codec byte is 160 and still it disconnects On Thu, Mar 11, 2010 at 12:03 AM, Omotayo adefilabi...@gmail.commailto:adefilabi...@gmail.com wrote: Hello, When i call from hq sccp phone to cme sip phone, it rings but when i pick up. it disconnects also when i call from cme sip phone to hq (sccp and sip) phone it rings on the hq phones when i pick t disconnects and contnues ringing on the sip phone i have a transcoder configured on the trunk Any one with a fix thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] IPexpert Bootcamp, May 2010, San Jose
I am going to attend the IPexpert five day bootcamp in San Jose this coming May. Would anyone be interested in splitting a hotel room to save some money? Please let me know soon as I will probably start looking for deals this week. I will also do the mock labs in June. If you'll be at that one let me know and we can work something out for a room split as well. Matthew Berry 612-424-5044 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Theory questions from Vol 1, Labs 5a-c
All - I have several theory questions that came up during my time spent in labs 5a, 5b, and 5c. I am hoping that some super smart CCIE candidates may have found the answer to these questions. Roger and Ossamah, I'm looking to you guys to represent. :) 1. How do you source RAS messages from a specific port on a gateway? I know you can define the h225 listen-port but that seems to only be one part. You can also configure the gateway to talk to the gatekeeper over a specific port using h323-gateway voip ip PL ipaddr 10.10.110.1 1719. How could I tell the gatekeeper to use a different port for interacting with RAS messages? 2. Registering SCCP full qualified E.164 DNs can cause issues with BACD and CUE voicemail pilots. Why is that? What if the lab requires us to register all CUCME phone DNs with the gatekeeper? 3. What is the purpose of application dial rules? Lab 5c utilized it for Mobile Connect and Mobile Voice Access, but there was no explanation on the Proctor Guide or the volume one walkthroughs. Can someone speak to the value there? It seems like it would allow a user to enter their own remote destination numbers (via ccmuser) without the access code of 9 and the dial rule could automatically prefix the 9. Thoughts? 4. When it comes to mobility,I know that I can associate an end user with a phone via End Users. I can also set an Owner User ID via the Device page. I can also associate the user with a line. What is the big difference between these features and what account association is required for what service? Those are my questions for now. :) ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Cisco Live 2010
Would anyone be willing to bring a microphone and record Ben's 8 hour techtorial? - Sent from my Blackberry From: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com To: OSL Group ccie_voice@onlinestudylist.com Sent: Tue Mar 02 12:31:19 2010 Subject: [OSL | CCIE_Voice] Cisco Live 2010 Anyone else attending Cisco Live this year in Vegas? I'm looking through the course list and trying to determine what would be the most appropriate sessions relative to the lab and it seems the UC related sessions have matured / grown in quantity since I went two years ago. Has anyone here gone to Ben Ng's 8hr Techtorial? I'm considering going to it, and would love any input anyone has on how beneficial it is to attend that. I don't think I need to spend the $$ on an 8hr lab with Proctor Labs available, and I'm looking at the four hour + dialing lab as an alternative option. If you're going and you haven't registered, consider using this link and sending me some Cisco Store bucks ;-) http://www.ciscolive2010.com/portal/registration/1267553946442 -Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Failure of Adding subscriber-VM Server 2.0
Good point, Roger. CUCM Pub needs something to validate against. - Sent from my Blackberry From: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com To: ShinGei Yong shingei.y...@gmail.com; ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Mon Mar 01 04:27:15 2010 Subject: Re: [OSL | CCIE_Voice] Failure of Adding subscriber-VM Server 2.0 Hi Shingei, This maybe a long shot, but anyway, have you added the sub as a UCM server on the pub before you tried to add it as a 2nd node? Roger Källberg Unified Communication Consultant Cygate AB From: ShinGei Yong [mailto:shingei.y...@gmail.com] Sent: den 1 mars 2010 03:29 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Failure of Adding subscriber-VM Server 2.0 Hi All, Im currently having the issue of adding the 2nd node to the UCM cluster. The error message as per attached jpg Configuration Validation with CUCM-PUB Failed. I've did some research based on the error message, either from CISCO or some older post. What i've tried out was: 1. Adding a NTP server to CUCM-PUB. CUCM-PUB was able to get the time source from my WINXP modified NTP server, and can be viewed from CUCM CLI. 2. Changed of Security Password. Based on CISCO explanation, it could be due to the security password mismatch between CUCM-PUB and SUB. I've reset the security password and reboot the CUCM-PUB, but still got no luck. I did tried to input an incorrect Security Password while adding the 2nd node, it did correctly prompt that the password was error. So confirmed that the security password entered was correct. 3. Some post mentioned that the MTU for the CUCM, will have some issue if leave it to default, but i'm not too sure what does it mean? Should i change it to MTU size to 1492? I'm currently using VMWare Server 2.0. regards, Shingei ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Adding Gatekeeper to CUCM Failure
Easy workaround is to use a different interface or, even better, setup another loopback. - Sent from my Blackberry From: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com To: Wael Agina waelag...@gmail.com; OSL Group ccie_voice@onlinestudylist.com Sent: Mon Mar 01 04:19:28 2010 Subject: Re: [OSL | CCIE_Voice] Adding Gatekeeper to CUCM Failure You can't have a H.323 gw and a GK registered on the same address in UCM. Roger Källberg Unified Communication Consultant Cygate AB From: Wael Agina [mailto:waelag...@gmail.com] Sent: den 1 mars 2010 10:18 To: OSL Group Subject: Re: [OSL | CCIE_Voice] Adding Gatekeeper to CUCM Failure Dear Team, I removed the gateway which has the same ip and i could add the gatekeeper with same ip. However i tried to add the h323 gateway again using same ip as gatekeeper , but it failed and give same error below Status [https://10.10.210.10:8443/ccmadmin/themes/VtgBlaf/Stop24.gif] Update failed. One of the required fields on the page has the same value as an entry that already exists in the database. Please check the corresponding Find List page to verify your entry does not exist. Any explination, Idea ? Regards, Wael Agina On Mon, Mar 1, 2010 at 12:12 PM, Wael Agina waelag...@gmail.commailto:waelag...@gmail.com wrote: Dear All, I am tring to add HQ as a gatekeeper to CUCM Pub but addition failed as below message: Status [https://10.10.210.10:8443/ccmadmin/themes/VtgBlaf/Stop24.gif] Add failed. One of the required fields on the page has the same value as an entry that already exists in the database. Please check the corresponding Find List page to verify your entry does not exist. Actually there is no gatekeeper added before , this is the first one ? SO find gatekeeper doesnt return anything. Any idea ? Note: this is the HQ RTR which is used also as H323 GW as per vol 2 lab 2 PG with same source IP / LO0 IP 10.10.100.1 Thanks and Best Regards, Wael Agina -- Thanks and Best Regards, Wael Agina ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 4A problems- All equipment is in error when loading initial configs
I had a similar issue yesterdat, Randall. When I tried to revert, it wouldn't go back to the original version. Although, I don't fully understand how PLs revert command works. - Sent from my Blackberry - Original Message - From: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Sun Feb 28 23:36:35 2010 Subject: [OSL | CCIE_Voice] Lab 4A problems- All equipment is in error when loading initial configs HI, I am getting errors loading all the equipment except br2. I have reverted and loaded the whole POD twice What do I do??? Randall -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Sunday, February 28, 2010 10:46 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 48, Issue 161 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: 7961 Factory Reset now Stuck in Upgrading (Roger K?llberg) 2. Re: 7961 Factory Reset now Stuck in Upgradin (Kamran Ahsanullah) 3. Re: [cisco-voip] SIP SK phones show different time in CME (Jason Aarons (US)) 4. Re: 7961 Factory Reset now Stuck in Upgradin (CCIETalk.com) -- Message: 1 Date: Sun, 28 Feb 2010 18:36:39 +0100 From: Roger K?llberg roger.kallb...@cygate.se Subject: Re: [OSL | CCIE_Voice] 7961 Factory Reset now Stuck in Upgrading To: CCIETalk.com cciet...@gmail.com, Jeff Garvas j...@cia.net Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Message-ID: 79fa99add19eda4c9880d26d736e50ef2d2cbbb...@ex2-sth.domain.root Content-Type: text/plain; charset=utf-8 Try this, Cisco 7965 SCCP to SIP Firmware Upgrade November 17th, 2009 by Mark Recently I needed to change the firmware on some Cisco 7965 phones from SCCP to SIP. By far the simplest method is loading the COP file on UCM and letting the phone upgrade on its own. In my case, this upgrade was being done without using UCM. The Cisco read-me doc for the SIP firmware covers the COP upgrade procedure only. It tells you that you may unzip the files on a TFTP server but there is no procedure which explains what else you must do to load the SIP firmware. In this example I am upgrading Cisco 7965 phones to SIP firmware 8.5. Once you have downloaded the zipped version of the SIP firmware from CCO place the unzipped files in your TFTP servers root directory. Modify your XMLDefaults.cnf.xml file so the load information matches your firmware. loadInformation8 model=?Cisco 7965?SIP45.8-5-3TH1/loadInformation8 You should connect your IP phone to LAN where DHCP provides the IP, subnet, and TFTP server IP. Make sure your phone has DHCP enabled = YES. Your DHCP server needs to support DHCP Options. TFTP option 66 is required for Cisco phones running SIP. Option 66 can be used to provide an IP address (recommended) but can also support a DNS names (assuming you are also providing at least one DNS server IP via DHCP). Option 150 only supports IP addresses and is required for SCCP firmware. You can safely configure your DHCP to issue both TFTP options. Next pull the power from your phone and plug it back in. Hold down # until the line keys start to blink and press 123456789*0# and your phone should reset. Your phone should display ?Upgrading? on the screen. If you are using a Unix based tftp server you can execute tcpdump port 69 and you should see your phone requesting the files. Your phone should display the progress of the SIP firmware upgrade and eventually reboot. After it reboots you can press Settings Model Information and scroll down until you see the Call Control Protocol = SIP. If you performed a factory reset and did not have DHCP enabled then your phone is most likely stuck at the Upgrading screen. Pressing keys on the phone will not change the status. At this point you should pull the power, plug it back in, hold # and then enter the keys 3491672850*# to factory reset the phone. This allows the phone to clear its flash and still download new firmware. Your screen is going to be totally black and it will appear as if your phone is not functional, but the phone is really sending a DHCP request and waiting for an IP, subnet, and TFTP IP assignment before proceeding to download the firmware. All of this is happening
[OSL | CCIE_Voice] IP-to-IP Gateway Question
I was going though Mark Snow's VoD for v3. In the call routing video, Mark touches on IP-to-IP gateway functionality, but I felt there was quite a bit left out. It didn't seem to complete to me. One of the questions that came out of watching that video is what is the big difference in features between a normal gateway and a licensed CUBE? Cisco gives an example configuration of a CUBE on their website at http://www.cisco.com/en/US/products/sw/voicesw/ps5640/products_configuration_example09186a00808ead0f.shtml (see below). All of these commands I can do on a gateway without the CUBE license. So what are you paying for when you get the CUBE license? voice service voip allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip allow-connections h323 to h323 dial-peer voice 1 voip session target ipv4:10.13.8.150 incoming called-number 8... dtmf-relay h245-alphanumeric codec g711ulaw ! dial-peer voice 2 voip destination-pattern 8... session protocol sipv2 session target ipv4:10.13.8.16 dtmf-relay rtp-nte codec g711ulaw Digital Footprint: Skype: ciscovoiceguru ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] IP-to-IP Gateway Question
That's good to know. Pursuing a CCIE is almost as much a crash-course in the disjointed way that Cisco utilizes licensing and hides documents on their website. :) From: Roger Källberg [mailto:roger.kallb...@cygate.se] Sent: Tuesday, February 23, 2010 7:27 AM To: Berry, Matthew J.; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] IP-to-IP Gateway Question Hi Matthew, There is no difference, just as there isn't any difference for the GK feature up to the IOS version were this became a licensed feature. I got a mail a while back from Vik with that version, it's post 12.4(15)T8 that you won't have the GK without the proper license. But as of now this isn't the case with the CUBE, not that I'm aware of anyway. So to put it short, it's just your conscience that stops you from using features you haven't paid for ;-) Roger Källberg Unified Communication Consultant Cygate AB From: Berry, Matthew J. [mailto:mjbe...@krollontrack.com] Sent: den 23 februari 2010 12:28 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] IP-to-IP Gateway Question I was going though Mark Snow's VoD for v3. In the call routing video, Mark touches on IP-to-IP gateway functionality, but I felt there was quite a bit left out. It didn't seem to complete to me. One of the questions that came out of watching that video is what is the big difference in features between a normal gateway and a licensed CUBE? Cisco gives an example configuration of a CUBE on their website at http://www.cisco.com/en/US/products/sw/voicesw/ps5640/products_configuration_example09186a00808ead0f.shtml (see below). All of these commands I can do on a gateway without the CUBE license. So what are you paying for when you get the CUBE license? voice service voip allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip allow-connections h323 to h323 dial-peer voice 1 voip session target ipv4:10.13.8.150 incoming called-number 8... dtmf-relay h245-alphanumeric codec g711ulaw ! dial-peer voice 2 voip destination-pattern 8... session protocol sipv2 session target ipv4:10.13.8.16 dtmf-relay rtp-nte codec g711ulaw Digital Footprint: Skype: ciscovoiceguru ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUC Not Licensed For VPIM
Is there a plan to add that? Ben said it could be a testable topic. - Sent from my Blackberry From: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com To: scott carruthers scarruthe...@hotmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Sun Feb 21 12:55:56 2010 Subject: Re: [OSL | CCIE_Voice] CUC Not Licensed For VPIM Hi Scott, Currently Proctorlabs gear is not licensed for VPIM, so we cannot make test with the feature for the time being, On Sat, Feb 20, 2010 at 7:46 PM, scott carruthers scarruthe...@hotmail.commailto:scarruthe...@hotmail.com wrote: When I attempt to add a VPIM location is Unity Connection I receive the following license error. Are the proctorlabs servers not licensed for VPIM? Anyone attempt VPIM in these labs yet? Status The requested operation would result in a license violation. Unable to create VPIM Location Save New VPIM Location Display Name* Dtmf Access ID* Partition cuc7-pub Partition Domain Name* IP Address* Remote phone prefix Save Fields marked with an asterisk (*) are required. The Demo license info show nothing for VPIM: SERVER this_host ANY VENDOR cisco INCREMENT LicVoicePortsMax cisco 7.0 permanent 2 HOSTID=ANY \ NOTICE=LicFileIDFOR_DEMO_ONLY.lic/LicFileIDLicLineID0/LicLineID \ PAKdummyPak/PAK SIGN=A3DF5BBED8B0 INCREMENT LicSubscribersMax cisco 7.0 permanent 10 HOSTID=ANY \ NOTICE=LicFileIDFOR_DEMO_ONLY.lic/LicFileIDLicLineID1/LicLineID \ PAKdummyPak/PAK SIGN=FA226A483396 INCREMENT LicVMISubscribersMax cisco 7.0 permanent 10 HOSTID=ANY \ NOTICE=LicFileIDFOR_DEMO_ONLY.lic/LicFileIDLicLineID2/LicLineID \ PAKdummyPak/PAK SIGN=22D6A4F63854 INCREMENT LicAdvancedUserMax cisco 7.0 permanent 10 HOSTID=ANY \ NOTICE=LicFileIDFOR_DEMO_ONLY.lic/LicFileIDLicLineID3/LicLineID \ PAKdummyPak/PAK SIGN=85B5BD2CDF32 INCREMENT LicRealspeakSessionsMax cisco 7.0 permanent 2 HOSTID=ANY \ NOTICE=LicFileIDFOR_DEMO_ONLY.lic/LicFileIDLicLineID4/LicLineID \ PAKdummyPak/PAK SIGN=24848F662AEC INCREMENT LicServerBackend cisco 7.0 permanent 1 HOSTID=ANY \ NOTICE=LicFileIDFOR_DEMO_ONLY.lic/LicFileIDLicLineID5/LicLineID \ PAKdummyPak/PAK SIGN=6750CF4C26B4 INCREMENT LicIMAPSubscribersMax cisco 7.0 permanent 10 HOSTID=ANY \ NOTICE=LicFileIDFOR_DEMO_ONLY.lic/LicFileIDLicLineID6/LicLineID \ PAKdummyPak/PAK SIGN=0A5E3C90C67A INCREMENT LicUnityVoiceRecSessionsMax cisco 7.0 permanent 2 \ HOSTID=ANY \ NOTICE=LicFileIDFOR_DEMO_ONLY.lic/LicFileIDLicLineID7/LicLineID \ PAKdummyPak/PAK SIGN=12E962E6B592 INCREMENT LicServerVoiceRec cisco 7.0 permanent 1 HOSTID=ANY \ NOTICE=LicFileIDFOR_DEMO_ONLY.lic/LicFileIDLicLineID8/LicLineID \ PAKdummyPak/PAK SIGN=5C6FF1C641AE INCREMENT LicMaxMsgRecLenIsLicensed cisco 7.0 permanent 1 HOSTID=ANY \ NOTICE=LicFileIDFOR_DEMO_ONLY.lic/LicFileIDLicLineID9/LicLineID \ PAKdummyPak/PAK SIGN=573BA6B413B6 INCREMENT LicRegionIsUnrestricted cisco 7.0 permanent 1 HOSTID=ANY \ NOTICE=LicFileIDFOR_DEMO_ONLY.lic/LicFileIDLicLineID10/LicLineID \ PAKdummyPak/PAK SIGN=40EBACAE87D8 Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now.http://clk.atdmt.com/GBL/go/201469229/direct/01/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Calling Party Numbering Plan
All - I did Vol 1 Lab 5a yesterday and ran into a question about Calling Party Numbering Plan. I have pasted Cisco's explanation below, but I'm looking for some insight as to why we would ever use this setting and how I would know if the lab was trying to validate my understanding of this concept. Any ideas? · Calling Party Numbering Plan: Choose the format for the numbering plan in calling party directory numbers. oCisco Unified Communications Manager sets the calling DN numbering plan. Cisco recommends that you do not change the default value unless you have advanced experience with dialing plans such as NANP or the European dialing plan. You may need to change the default in Europe because Cisco Unified Communications Manager does not recognize European national dialing patterns. You can also change this setting when you are connecting to PBXs by using routing as a non-national type number. oChoose one of the following options: § Cisco Unified Communications Manager-Use when the Cisco Unified Communications Manager sets the Numbering Plan in the directory number. § ISDN-Use when you are dialing outside the dialing plan for your country. § National Standard-Use when you are dialing within the dialing plan for your country. § Private-Use when you are dialing within a private network. § Unknown-Use when the dialing plan is unknown. Thanks, Matthew Berry, Sr. Unified Communications Engineer, CCVP Kroll Ontrack | 9023 Columbine Road, Eden Prairie, MN 55347 +1 952 516 3748 | Fax +1 952 516 3646 | Mobile +1 612 836 7626| mjbe...@krollontrack.commailto:agutz...@krollontrack.com www.krollontrack.comhttp://www.krollontrack.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Calling Party Numbering Plan
Jeff - Thanks for your reply. I understand the use of calling part umber types like Subscriber, National, and International to globalize calls as they egress a gateway. What I don't understand is the user of the numbering plan. From Cisco's verbiage, it seems that this may never be used until you're dealing with a cluster in Europe or other place with a screwy national dial plan. We were asked to see the plan to ISDN for a task in Vol 1 5a, which is why I asked. Anyone have an explanation? Thanks, Matthew Berry Office +1 952 516 3748 | Mobile +1 612 836 7626| mjbe...@krollontrack.commailto:agutz...@krollontrack.com From: Jeff Knuckle [mailto:jknuc...@nationwidelab.com] Sent: Monday, February 15, 2010 4:06 PM To: Berry, Matthew J.; ccie_voice@onlinestudylist.com Subject: RE: Calling Party Numbering Plan To answer the first part of your question, Calling party numbering Plan would be use to localize global (Calling) numbers on egress calls. http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/dialplan.html#wp1172104 Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Berry, Matthew J. Sent: Monday, February 15, 2010 3:43 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Calling Party Numbering Plan All - I did Vol 1 Lab 5a yesterday and ran into a question about Calling Party Numbering Plan. I have pasted Cisco's explanation below, but I'm looking for some insight as to why we would ever use this setting and how I would know if the lab was trying to validate my understanding of this concept. Any ideas? · Calling Party Numbering Plan: Choose the format for the numbering plan in calling party directory numbers. oCisco Unified Communications Manager sets the calling DN numbering plan. Cisco recommends that you do not change the default value unless you have advanced experience with dialing plans such as NANP or the European dialing plan. You may need to change the default in Europe because Cisco Unified Communications Manager does not recognize European national dialing patterns. You can also change this setting when you are connecting to PBXs by using routing as a non-national type number. oChoose one of the following options: § Cisco Unified Communications Manager-Use when the Cisco Unified Communications Manager sets the Numbering Plan in the directory number. § ISDN-Use when you are dialing outside the dialing plan for your country. § National Standard-Use when you are dialing within the dialing plan for your country. § Private-Use when you are dialing within a private network. § Unknown-Use when the dialing plan is unknown. Thanks, Matthew Berry, Sr. Unified Communications Engineer, CCVP Kroll Ontrack | 9023 Columbine Road, Eden Prairie, MN 55347 +1 952 516 3748 | Fax +1 952 516 3646 | Mobile +1 612 836 7626| mjbe...@krollontrack.commailto:agutz...@krollontrack.com www.krollontrack.comhttp://www.krollontrack.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Vol 1 - Lab 5 - Task 5.5 / Setting up TEHO for 212 calls from BR1 through HQ H.323 GW
Vol 1 - Lab 5 - Task 5.5 / Setting up TEHO for 212 calls from BR1 through HQ H.323 GW I can get TEHO to work when dialing a 617 area code number from HQ Phone 2, routing the call over the WAN, out the BR1 MGCP gateway. It works like a charm. It appends the + which seems to come from the 9.1617XXX translation pattern in PT-HQ-PSTN. Problem: I cannot get the + to be sent out when setting up TEHO for 212 area code calls from BR1 through HQ's H.323 GW. All of my settings for the BR1 site are identical to the HQ site. My only guess is that TEHO over WAN and out the BR1 MGCP gateway is using MGCP and not H.323. I can append a + using a dial-peer on the H.323 gateway, but I'm not sure if that is the best way to do it. It seems like Ben was saying that however you produce the end results in the lab is all that matters. What do you guys think? Am I missing something? Digital Footprint: Skype: ciscovoiceguru ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] DS0 Port Utilization
All - This is somewhat off-topic for the CCIE Voice lab... Well, it is actually completely off-topic. Is anyone here using a SNMP-based tool to monitor in real-time the DS0 usage of your PRIs? We have been using PRTG enterprise for our data traffic. However, I'm looking for something to monitor my circuits. Thanks, Matthew Berry, Sr. Unified Communications Engineer, CCVP Kroll Ontrack | 9023 Columbine Road, Eden Prairie, MN 55347 952 516 3748 | Fax 952 516 3646 | Mobile 612 836 7626| mjbe...@krollontrack.commailto:agutz...@krollontrack.com www.krollontrack.comhttp://www.krollontrack.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol1 Lab 4A - X-Lite Issues
Steve, What codec work is going on between the sites? If you are calling from HQ to BR2, your region will be using G.729r8. However, you BR2 SIP phone is likely setup for G.711ulaw. That could be why when you answer the call, it drops. Codec negotiation could fail. Remember, SIP dial peers cannot deal with voice class codec commands. Even though it takes it, the dial-peer will not use it. You must hard code your codec selection into the dial peers and then use a transcoder to change the codec. Try messing around with the codecs and see if that changes anything. Thanks, Matthew Berry Office 952 516 3748 | Mobile 952 221 2814| mjbe...@krollontrack.commailto:agutz...@krollontrack.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Steve Denney (stdenney) Sent: Thursday, February 04, 2010 2:25 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Vol1 Lab 4A - X-Lite Issues Hitting an interesting problem and just wondering if anyone else has seen similar symptoms... Working on Vol1, Lab 4A, Task 4.5. This is the task where you set up a SIP Route Pattern and use SIP URI dialing to dial the X-Lite CME SIP Phone (BR2 Ph 4, DN 3006) from the CIPC SIP Phone (HQ Ph2, DN 5002). When dialing from 5002 to 3006 (using the corporate directory on CIPC, as shown in the lab), the X-Lite rings, but hangs up immediately after the call is answered. The output of debug ccsip mess is attached. Looks like the X-Lite is sending a SIP BYE message with the description of Illegal Sdp Negotiation. I tried a call in the other direction as well - direct dial from 3006 to 5002. The CIPC rings, but you cannot actually answer the call. The debug in this case shows a 503 Service Unavailable message, and the display on the X-Lite says Call failed: Service Unavailable. I've double and triple checked all configs (including allow-connections sip to sip), reloaded all routers, Googled for similar issues, and am now officially stumped. :) Debugs attached. Any ideas? cheers, steve ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] WAN topology in lab
Guys - From what Ben Ng told me, OSPF would be setup and not a part of troubleshooting. I think it's safe to say we can focus on what the blueprint says and the blueprint alone. If it doesn't say OSPF, then we can ignore OSPF. Thanks, Matthew Berry Office 952 516 3748 | Mobile 952 221 2814| mjbe...@krollontrack.commailto:agutz...@krollontrack.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Garvas Sent: Wednesday, February 03, 2010 8:48 AM To: Roger Källberg Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] WAN topology in lab Roger, Earlier in the discussion he answered someone else. He said he'd leave OSPF to his counterparts in RS, but that you could have misconfigured layer 2 issues that impact OSPF that you need to fix. Post #35 by Matthew Berry (From this list I think?) asked, and in post #36 ben answered around 6pm yesterday. However, your question might give us a more in-depth look at the scope of the areas. I couldn't get him to answer about the RTP scheduling issue. Looks like the written will be refreshed mid to late summer. -jeff 2010/2/3 Roger Källberg roger.kallb...@cygate.semailto:roger.kallb...@cygate.se Hi Jeff, thank you for your answer and suggestion to post the question on the Ask the Expert forum. I just did that, I'll update this post when Ben Ng has replyed. Brgds, Roger Källberg Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ Direkt: +46108787498 Växel: +46108787400 roger.kallb...@cygate.semailto:roger.kallb...@cygate.se Från: Jeff Garvas [j...@cia.netmailto:j...@cia.net] Skickat: den 2 februari 2010 23:15 Till: Roger Källberg Kopia: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Ämne: Re: [OSL | CCIE_Voice] WAN topology in lab I've been assuming that the OSPF would be built for you much as it is in the ipexpert environment since OSPF isn't really on the blueprint, so unless you break it yourself I'd expect it to be fully functional. Looking at section 1.xx of the blue print the only infrastructure you're supposedly tested on is vlans, dhcp, tftp and ntp. Granted they're all dependent upon OSPF, but I've never considered the underlying routing protocol to be part of the voice environment (for the lab that is). Roger: That might be a good question for the ask the expert on netpro that Ben Ng. is answering questions on through most of this month, but I'm assuming the answer is it shouldn't be a testable topic. 2010/2/2 Roger Källberg roger.kallb...@cygate.semailto:roger.kallb...@cygate.se Hi, I just reread the OSPF chapters in my old CCNP course material as a fresh up for my lab preparation and wonder if someone could tell me if we could, or should, expect to get a single area or a multiple area topology in the lab? Regards, Roger Källberg Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ Direkt: +46108787498 Växel: +46108787400 roger.kallb...@cygate.semailto:roger.kallb...@cygate.se ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUCM 7.0 DHCP Server
Wilson, There is a known bug in CUCM 7.0 (not sure if it's in 7.1) where CSA will disallow DHCP requests if you initially installed the CUCM software and did not configure DHCP during the install process. 1.If CSA is enabled, CUCM-facilitated DHCP may fail. You may need to disable CSA on CUCM: Utils csa status Utils csa disable [requires restart] Thanks, Matthew Berry Office 952 516 3748 | Mobile 952 221 2814| mjbe...@krollontrack.commailto:agutz...@krollontrack.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jason Granat Sent: Tuesday, February 02, 2010 1:09 PM To: wilson.sam...@usc-bt.com; 2xcci...@gmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CUCM 7.0 DHCP Server Is it in the same subnet as the DHCP clients (phones)? If it is in a different subnet do you have ip helper-address with the CUCM IP configured on the clients' default-gateway interface? From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of wilson.sam...@usc-bt.com Sent: Tuesday, February 02, 2010 10:53 AM To: 2xcci...@gmail.com; ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CUCM 7.0 DHCP Server Hi All, I know this should be incredibly easy, however for some reason my brain is not helping with me on this. I have installed a CUCM 7.1 Pub (no Sub) and want to make it a DHCP Server (as normally required by most of the labs), even after disabling the CSA, my phones are not taking any IP Address from the CUCM DHCP Server. Btw, the CUCM is on a VMWare VM and has a bridged network card with the host. I can ping everything and everything is fine if I use my ASA as the DHCP. However the moment I use the CUCM DHCP Server , the phones cant get IP Address from it. Anything I have been ignoring? Regards Wilson Samuel http://slash128.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP Phone Codec ?
According to Vik's vol 1 walkthrough, that willk not work. CUCME SIP phones support one codec only. - Sent from my Blackberry From: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com To: Mike Brooks 2xcci...@gmail.com Cc: OSL Group ccie_voice@onlinestudylist.com Sent: Mon Feb 01 20:28:55 2010 Subject: Re: [OSL | CCIE_Voice] SIP Phone Codec ? Hi Mike, Rather than hardcoding the codec under the voice register pool X use the voice-class codec X command in place of the codec for example; voice class codec 1 codec preference 1 g729r8 codec preference 2 g711ulaw voice register pool 1 id mac .. type 7961 voice-class codec 1 This should work, I haven't tested it yet :-) On Tue, Feb 2, 2010 at 12:55 PM, Mike Brooks 2xcci...@gmail.commailto:2xcci...@gmail.com wrote: Alright, this is an easy question. I just want to verify what I am thinking is correct. On CME the sip phones have a limitation of having to be hardcoded to a codec, for example G711 OR G729. I believe the fact that on CME the phones cannot negotiate codec is a limitation with CME and not the sip phones. When phones are registered with callmanager they do not have this problem. Please correct me if I am wrong. Thanks, Mike ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUCME SIP Issues
Hey Wayne – I really enjoy the Volume 1 Video Solutions. They are a great compliment to Workbook 1. Last Sunday, I went through exercise 3A and 3B. Just this morning, I finished watching the corresponding techtorials recorded by Vik. I’m impressed. When I got this new product, I wasn’t sure how much of a value add it was going to be, but I’m impressed by the additional insight Vik gives in the videos. I’ll keep you posted on my progress as the weeks and months progress. Thanks, Matthew Berry Office 952 516 3748 | Mobile 952 221 2814| mjbe...@krollontrack.commailto:agutz...@krollontrack.com From: Wayne Lawson [mailto:groupst...@ipexpert.com] Sent: Tuesday, January 26, 2010 9:53 PM To: Berry, Matthew J. Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CUCME SIP Issues Matthew - How are the Vol 1 Video Solutions working out? Keep in touch! Regards, Wayne A. Lawson II - CCIE #5244 Founder President - IPexpert Mailto: wlaw...@ipexpert.commailto:wlaw...@ipexpert.com Telephone: +1.810.326.1444, ext. 101 Live Assistance, Please visit: www.ipexpert.com/chathttp://www.ipexpert.com/chat eFax: +1.810.454.0130 ::Message sent from iPhone:: IPexpert is a premier provider of Classroom and Self-Study Cisco CCNA (RS, Voice Security), CCNP, CCVP, CCSP and CCIE (RS, Voice, Security Service Provider) Certification Training with locations throughout the United States, Europe and Australia. Be sure to check out our online communities at www.ipexpert.com/communitieshttp://www.ipexpert.com/communities and our public website at www.ipexpert.comhttp://www.ipexpert.com. On Jan 26, 2010, at 10:49 PM, Berry, Matthew J. mjbe...@krollontrack.commailto:mjbe...@krollontrack.com wrote: Scenario: I have an X-Lite softphone setup with a dn of 20004. I also setup another dn of 20005 to call forward all to 20004. The dn of 20005 is not assigned to another phone. In this scenario, there is only one phone registered to the CUCME SIP instance. Problem: I go off-hook, dial 2-0-0-0-5 and receive a Call failed: Not Acceptable Media Debug: SIP/2.0 488 Not Acceptable Media Via: SIP/2.0/UDP 10.25.3.210:55446;branch=z9hG4bK-d8754z-262a3d2cba0faf54-1---d8754z-;rport From: CUCME SIPsip:20...@10.25.3.200;tag=4c43d170 To: 20005sip:20...@10.25.3.200;tag=1B1F3408-1F39 Date: Wed, 27 Jan 2010 03:45:22 GMT Call-ID: M2JmNGU0OTZhM2Y4MzJhNTg2YmY5NDc4NmFlZjI2ZmU. Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow-Events: telephone-event Reason: Q.850;cause=65 Content-Length: 0 Question 1: Why does it give me the 488 error? Question 2: Do DNs need to be assigned to working phones in order for calls to be directed to them? If so, what happens if a SIP phone with said dn loses network connectivity? Thanks, Matthew Berry, Sr. Unified Communications Engineer, CCVP Kroll Ontrack | 9023 Columbine Road, Eden Prairie, MN 55347 952 516 3748 | Fax 952 516 3646 | Mobile 952 221 2814| mjbe...@krollontrack.commailto:agutz...@krollontrack.com www.krollontrack.comhttp://www.krollontrack.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME files for phones
Randall - Those four file types make up the firmware that the Cisco IP phone uses. The 7940/7960 phones use the .sb2 and .bin commands. Proctor Labs (ie. IP Expert) uses 7960s in their racks. However, the actual lab is going to use 7965s. If you look at the newer phone models, such as the 7965, they don't use the same images and file types as previous versions. Take a look at the CUCME SRND: http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeinstl.html#wp1070512 The following example shows a list of phone firmware files that are installed in flash memory for the Cisco Unified IP Phone 7911: tftp-server flash:SCCP11.7-2-1-0S.loads tftp-server flash:term06.default.loads tftp-server flash:term11.default.loads tftp-server flash:cvm11.7-2-0-66.sbn tftp-server flash:jar11.7-2-0-66.sbn tftp-server flash:dsp11.1-0-0-73.sbn tftp-server flash:apps11.1-0-0-72.sbn tftp-server flash:cnu11.3-0-0-81.sbn Here's an example of the firmware files used for the 7911 IP Phone. The format will be the same for the 7941, 7961, 7965, etc. Become familiar with this new file format because that's what you'll see on the lab. Hope this helps! Thanks, Matthew Berry Office 952 516 3748 | Mobile 952 221 2814| mjbe...@krollontrack.com -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Randall Crumm Sent: Tuesday, January 26, 2010 11:48 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME files for phones Hi, I want to know what are the four files in CME? I don't work with CME and I am looking at lab 3A .bin .loads .sb2 .sbn Thanks, Randall -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Tuesday, January 26, 2010 9:00 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 47, Issue 127 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Call Forward not working (Sivakumar Mahalingam) 2. Re: MULTICAST MOH over SIP TRUNK? (Vccie Vccie) -- Message: 1 Date: Tue, 26 Jan 2010 11:22:32 -0500 From: Sivakumar Mahalingam sima...@gmail.com Subject: [OSL | CCIE_Voice] Call Forward not working To: OSL Group ccie_voice@onlinestudylist.com Message-ID: 703b51d01001260822y1df82b73se07cd3463a9f3...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi All, I need some help for the below issue that am facing. I have CUCM 7.1.3(a) running on my VoIP network and the issue is ,when i setup forward all for Extn A to Extn B ,the off campus calls are forwarded to Extn B correctly and the on campus calls are not being forwarded and it rings the Extn A phone directly. If anyone of you have faced a simillar kind of problem,please let me know you thoughts. Thanks, Simah. -- next part -- An HTML attachment was scrubbed... URL: http://onlinestudylist.com/pipermail/ccie_voice/attachments/20100126/da62b3f3/attachment-0001.htm -- Message: 2 Date: Tue, 26 Jan 2010 10:29:43 -0600 From: Vccie Vccie voiceccie2...@gmail.com Subject: Re: [OSL | CCIE_Voice] MULTICAST MOH over SIP TRUNK? To: ccie_voice@onlinestudylist.com Message-ID: 8adf63bc1001260829x19043ec8kde845eed89336...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 After further testing I have confirmed that is indeed the IOS version that is not allowing for the Multicast MOH over a SIP trunk to a PSTN Termination. Tested versions: c2801-adventerprisek9_ivs-mz.124-20.T4.bin = PSTN MOH DOESNT WORK c2801-adventerprisek9_ivs-mz.124-22.T.bin = PSTN MOH WORKS Typology SIP/SKINNYPHONE - UCM (Multicast-PUB/MGRL) - SIPTRUNK - (HQ-2801 sourced Multicast MOH) PRI - (PSTN-2821) -PSTNPHONE If any one know anything to the contrary to my findings please respond as I am under the assumption that this is the final outcome. Thank you -- next part -- An HTML attachment was scrubbed... URL: http://onlinestudylist.com/pipermail/ccie_voice/attachments/20100126/97cae30a/attachment-0001.htm -- ___ CCIE_Voice mailing list CCIE_Voice@onlinestudylist.com http://onlinestudylist.com/mailman/listinfo/ccie_voice End of CCIE_Voice Digest, Vol 47, Issue 127
[OSL | CCIE_Voice] CUCME SIP Issues
Scenario: I have an X-Lite softphone setup with a dn of 20004. I also setup another dn of 20005 to call forward all to 20004. The dn of 20005 is not assigned to another phone. In this scenario, there is only one phone registered to the CUCME SIP instance. Problem: I go off-hook, dial 2-0-0-0-5 and receive a Call failed: Not Acceptable Media Debug: SIP/2.0 488 Not Acceptable Media Via: SIP/2.0/UDP 10.25.3.210:55446;branch=z9hG4bK-d8754z-262a3d2cba0faf54-1---d8754z-;rport From: CUCME SIPsip:20...@10.25.3.200;tag=4c43d170 To: 20005sip:20...@10.25.3.200;tag=1B1F3408-1F39 Date: Wed, 27 Jan 2010 03:45:22 GMT Call-ID: M2JmNGU0OTZhM2Y4MzJhNTg2YmY5NDc4NmFlZjI2ZmU. Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow-Events: telephone-event Reason: Q.850;cause=65 Content-Length: 0 Question 1: Why does it give me the 488 error? Question 2: Do DNs need to be assigned to working phones in order for calls to be directed to them? If so, what happens if a SIP phone with said dn loses network connectivity? Thanks, Matthew Berry, Sr. Unified Communications Engineer, CCVP Kroll Ontrack | 9023 Columbine Road, Eden Prairie, MN 55347 952 516 3748 | Fax 952 516 3646 | Mobile 952 221 2814| mjbe...@krollontrack.commailto:agutz...@krollontrack.com www.krollontrack.comhttp://www.krollontrack.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] IPExpert's Volume 1 Video Walkthroughs
All - Last week, I received the new IP Expert Volume 1 video walkthroughs that were recorded by Vik. Today is my labbing day so I decided to pop in the DVD and listen to it after each lab was finished. WOW! That's what I have to say. Vik brought up a ton of stuff that I otherwise would not have caught in the exercises. He's great to listen to and there's no doubt that he understands his stuff. I've gone through his walkthroughs of labs 1a-3a so far. I just wanted to send out a quick review of what I've experienced so far. I know there has been some discussion on the OSL about this new product. Let me vouch for it. It's excellent material. Digital Footprint: Skype: ciscovoiceguru ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Recommended Version of CME for CCIE Voice Lab
Cristobal mentioned that he is using CME 7.1.1.0 in his lab. According to the lab blueprint, it seems that the version that will be on the lab is CME 7.0. Is this correct? Are minor releases supported in the labs? For example, could we be tested on CUCM 7.1(2) and the new features that are added in this release? Otto/Vik, could you respond to this one? Thanks, Matthew Berry Office 952 516 3748 | Mobile 952 221 2814| mjbe...@krollontrack.commailto:agutz...@krollontrack.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cristobal Priego Sent: Friday, January 22, 2010 11:46 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CCME and SNR Hello all, I have a little issue here, hopefully you can help me out I have CME 7.1.1.0 installed and I have SNR on a few extensions configured working great, my problem starts when a call is connected on an extension that has SNR enabled, when i press the mobility soft key to transfer the call to my remote destination device (my cell phone) nothing happens i can't transfer the call. also as soon as i press the mobility soft key this is what i see on cme cli 000918: Jan 22 17:41:36.099: fStationSoftKeyEventMessage 6316 unknown press mobility key 37 could you point me on the right direction to resolve this thank you ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCME and SNR
Will CME 7.1 be in the lab? I thought 7.0 was on the blueprint. - Sent from my Blackberry From: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Fri Jan 22 11:46:23 2010 Subject: [OSL | CCIE_Voice] CCME and SNR Hello all, I have a little issue here, hopefully you can help me out I have CME 7.1.1.0 installed and I have SNR on a few extensions configured working great, my problem starts when a call is connected on an extension that has SNR enabled, when i press the mobility soft key to transfer the call to my remote destination device (my cell phone) nothing happens i can't transfer the call. also as soon as i press the mobility soft key this is what i see on cme cli 000918: Jan 22 17:41:36.099: fStationSoftKeyEventMessage 6316 unknown press mobility key 37 could you point me on the right direction to resolve this thank you ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Issues Configuring SCCP UCME 7.0
I did figure this out. Once I got the correct files in the flash, everything seemed to work. Thanks, Matthew Berry Office 952 516 3748 | Mobile 952 221 2814| mjbe...@krollontrack.commailto:agutz...@krollontrack.com From: Otto Sanchez [mailto:o...@ipexpert.com] Sent: Friday, January 22, 2010 12:55 PM To: Berry, Matthew J. Cc: vccie2010; OSL Group; Vik Malhi Subject: Re: [OSL | CCIE_Voice] Issues Configuring SCCP UCME 7.0 Matthew, Did you solve this issue? Is the tftp file being generated? do a sh telephony-service tftp and send us the results, would you also please send us a sh run, I also noticed you are using cme 7.1 (c2800nm-adventerprisek9-mz.124-24.T2.bin), and phone loads for cme 7.0, was that make on purpose? BR, On Thu, Jan 21, 2010 at 12:50 PM, Berry, Matthew J. mjbe...@krollontrack.commailto:mjbe...@krollontrack.com wrote: SCCP phones should register without an ephone-dn. I know from experience last week that SIP phones will not register without a DN. I am still trying to work through this issue. I've noticed that when I run debug tftp packets I get the following output about a file not being found: Jan 21 17:09:44.537: TFTP: Finished system:/its/XMLDefault.cnf.xml, time 00:00:00 for process 200 Jan 21 17:10:24.939: TFTP: Server request for port 49935, socket_id 0x491D35A4 for process 200 Jan 21 17:10:24.939: TFTP: read request from host 10.38.4.123(49935) via GigabitEthernet0/0 Jan 21 17:10:24.939: TFTP: Looking for CTLSEP00235E17AB31.tlv Jan 21 17:10:24.939: TFTP: Sending error 1 No such file Jan 21 17:10:25.131: TFTP: Server request for port 49936, socket_id 0x491D35A4 for process 200 Jan 21 17:10:25.135: TFTP: read request from host 10.38.4.123(49936) via GigabitEthernet0/0 Jan 21 17:10:25.135: TFTP: Looking for SEP00235E17AB31.cnf.xml Jan 21 17:10:25.135: TFTP: Sending error 1 No such file Jan 21 17:10:25.215: TFTP: Server request for port 49937, socket_id 0x491D35A4 for process 200 Jan 21 17:10:25.215: TFTP: read request from host 10.38.4.123(49937) via GigabitEthernet0/0 Jan 21 17:10:25.215: TFTP: Looking for XMLDefault.cnf.xml Jan 21 17:10:25.215: TFTP: Opened system:/its/XMLDefault.cnf.xml, fd 7, size 2740 for process 200 Thanks, Matthew Berry Office 952 516 3748 | Mobile 952 221 2814| mjbe...@krollontrack.commailto:agutz...@krollontrack.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Issues Configuring SCCP UCME 7.0
All - It'd be great if someone could comment on this issue I'm encountering. This is the first time I've setup UCME 7.0 (as compared to CME 3.x, 4.x). So I transferred the SCCP files for a 7961 phone to the root of flash. This is different than the IP Expert Appendix A documentation and Mark Snow's VoD CUCME lectures. They specify something called Type 3 file structure that places phone firmware in flash:PHONE/... The only reason I placed the firmware in the root of flash is based off other people's configs that have been shared on the OSL. When the phone contacts the TFTP server on my UCME router, it updates its firmware, but never registered. I have auto-reg-ephone configured and even manually configured the ephone in the config. However, it will not register. Does anyone have an idea of what's going on? I have debug tftp events and debug ephone register enabled. These is the output I get: Jan 21 14:33:44.682: TFTP: Looking for CTLSEP0021D8BB4A63.tlv Jan 21 14:33:44.822: TFTP: Looking for SEP0021D8BB4A63.cnf.xml Jan 21 14:33:45.054: TFTP: Opened system:/its/vrf1/XMLDefault7961.cnf.xml, fd 8, size 1199 for process 301 Jan 21 14:33:45.058: TFTP: Finished system:/its/vrf1/XMLDefault7961.cnf.xml, time 00:00:00 for process 301 Jan 21 14:33:46.238: TFTP: Looking for English_United_States/mk-sccp.jar Jan 21 14:33:46.378: TFTP: Looking for United_States/g3-tones.xml Jan 21 14:33:57.206: TFTP: Looking for CTLSEP00235E17AB31.tlv Jan 21 14:33:57.394: TFTP: Looking for SEP00235E17AB31.cnf.xml Jan 21 14:33:57.478: TFTP: Looking for XMLDefault.cnf.xml Router#show flash: -#- --length-- -date/time-- path 1 59455672 Nov 23 2009 20:39:38 c2800nm-adventerprisek9-mz.124-24.T2.bin ... 21 2494499 Jan 20 2010 23:35:52 apps41.8-3-2-27.sbn 22 547146 Jan 20 2010 23:35:56 cnu41.8-3-2-27.sbn 23 2452629 Jan 20 2010 23:36:06 cvm41sccp.8-3-2-27.sbn 24 530601 Jan 20 2010 23:36:10 dsp41.8-3-2-27.sbn 25 315827 Jan 20 2010 23:36:12 jar41sccp.8-3-2-27.sbn 26 638 Jan 20 2010 23:36:12 SCCP41.8-3-3S.loads 27 642 Jan 20 2010 23:36:12 term41.default.loads 28 642 Jan 20 2010 23:36:14 term61.default.loads Router#show run tftp-server flash:apps41.8-3-2-27.sbn alias apps41.8-3-2-27.sbn tftp-server flash:cnu41.8-3-2-27.sbn alias cnu41.8-3-2-27.sbn tftp-server flash:cvm41sccp.8-3-2-27.sbn alias cvm41sccp.8-3-2-27.sbn tftp-server flash:dsp41.8-3-2-27.sbn alias dsp41.8-3-2-27.sbn tftp-server flash:jar41sccp.8-3-2-27.sbn alias jar41sccp.8-3-2-27.sbn tftp-server flash:SCCP41.8-3-3S.loads alias SCCP41.8-3-3S.loads tftp-server flash:term41.default.loads alias term41.default.loads tftp-server flash:term61.default.loads alias term61.default.loads ! telephony-service max-ephones 3 max-dn 10 ip source-address 192.168.100.1 port 2000 load 7961 SCCP41.8-3-3S max-conferences 8 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Jan 21 2010 14:23:04 ! ephone 1 device-security-mode none mac-address 0021.D8BB.4A63 type 7961 Thanks, Matthew Berry, Sr. Unified Communications Engineer, CCVP Kroll Ontrack | 9023 Columbine Road, Eden Prairie, MN 55347 952 516 3748 | Fax 952 516 3646 | Mobile 952 221 2814| mjbe...@krollontrack.commailto:agutz...@krollontrack.com www.krollontrack.comhttp://www.krollontrack.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Issues Configuring SCCP UCME 7.0
SCCP phones should register without an ephone-dn. I know from experience last week that SIP phones will not register without a DN. I am still trying to work through this issue. I've noticed that when I run debug tftp packets I get the following output about a file not being found: Jan 21 17:09:44.537: TFTP: Finished system:/its/XMLDefault.cnf.xml, time 00:00:00 for process 200 Jan 21 17:10:24.939: TFTP: Server request for port 49935, socket_id 0x491D35A4 for process 200 Jan 21 17:10:24.939: TFTP: read request from host 10.38.4.123(49935) via GigabitEthernet0/0 Jan 21 17:10:24.939: TFTP: Looking for CTLSEP00235E17AB31.tlv Jan 21 17:10:24.939: TFTP: Sending error 1 No such file Jan 21 17:10:25.131: TFTP: Server request for port 49936, socket_id 0x491D35A4 for process 200 Jan 21 17:10:25.135: TFTP: read request from host 10.38.4.123(49936) via GigabitEthernet0/0 Jan 21 17:10:25.135: TFTP: Looking for SEP00235E17AB31.cnf.xml Jan 21 17:10:25.135: TFTP: Sending error 1 No such file Jan 21 17:10:25.215: TFTP: Server request for port 49937, socket_id 0x491D35A4 for process 200 Jan 21 17:10:25.215: TFTP: read request from host 10.38.4.123(49937) via GigabitEthernet0/0 Jan 21 17:10:25.215: TFTP: Looking for XMLDefault.cnf.xml Jan 21 17:10:25.215: TFTP: Opened system:/its/XMLDefault.cnf.xml, fd 7, size 2740 for process 200 Thanks, Matthew Berry Office 952 516 3748 | Mobile 952 221 2814| mjbe...@krollontrack.commailto:agutz...@krollontrack.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCCX Issue
Are you changing the script name under the application in UCCX? Have you added the script under the RM subsystem? I think that's where it is although I may be thinking of IP-IVR. - Sent from my Blackberry From: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com To: Otto Sanchez o...@ipexpert.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Thu Jan 21 21:14:49 2010 Subject: Re: [OSL | CCIE_Voice] UCCX Issue Yes Otto, I did that please but still having same issue. On Thu, Jan 21, 2010 at 1:13 PM, Otto Sanchez o...@ipexpert.commailto:o...@ipexpert.com wrote: Did you validated the script once it was saved with the new name?, a common cause for this error is that the queue name is invalid in the application section configuration, On Thu, Jan 21, 2010 at 1:37 AM, vccie2010 vccie2...@gmail.commailto:vccie2...@gmail.com wrote: Well I am just saving the defualt ICD script as a different name icdtest and the moment I call I get the error message I posted ealrier. I have all csq etc as taken care of as with default script it warks fine. On Wed, Jan 20, 2010 at 6:50 PM, kill mill jha...@gmail.commailto:jha...@gmail.com wrote: THis is a general issue you have to decode the script to see what the issue is. plus check which script you are referecing and all the parameters csq etc are in line On Wed, Jan 20, 2010 at 8:45 PM, vccie2010 vccie2...@gmail.commailto:vccie2...@gmail.com wrote: I have UCCX on vmware. I am able to make calls succesfully to UCCX when there is default ICD script selected, but once I open the default ICD script in CRS editor and rename that suppose as icdnew and upload and select i, now when I call the trigger it prompts Thank you for calling…I am sorry. We are currently experiencing system problem. Please try again later Does anyone had same issue or can guide me what could be the problem. The CRS editor was donwloaded from the UCCX server itself. Seems like somehow the CRS editor is not saving the .aef file properly or ??? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.comhttp://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SCCP to SIP Firmware Upgrade
Vik - Thanks for the reply. I've done phone conversions in UCM before. They're pretty easy. My only concern is going into the lab and being told to perform the conversion on the CME router itself. For example, if they suggest that I upgrade to a certain version of firmware for SIP that exists on CME, but not on UCM. I was searching through the OSL, and saw you mentioned that a CME-based SCCP to SIP upgrade would not be on the exam. You said it was raised in the techtorial at networks last year. (ref: http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg12404.html ). Did I understand that correctly? Also, I am not seeing the CME-7-0-full-readme-v.1.0.txt file in the CME 7.0(1) zip file I downloaded from Cisco. Has it been removed from the Cisco site? In Mark Snow's v3 VoD, he said to use that file as the building blocks for your CME configuration in the lab. I'm wondering if that's even a possibility now if the file is not in the ZIP file. Thanks, Matthew Berry Office 952 516 3748 | Mobile 952 221 2814| mjbe...@krollontrack.commailto:agutz...@krollontrack.com From: Vik Malhi [mailto:vma...@ipexpert.com] Sent: Wednesday, January 20, 2010 12:09 AM To: Berry, Matthew J.; OSL Group Subject: Re: [OSL | CCIE_Voice] SCCP to SIP Firmware Upgrade I would use UCM to do the firmware upgrade. Follow these steps: (1) Add the CME phone into the UCM database (add a new phone). (2) Assign a number to the phone- dummy number will do, just necessary for SIP phones to register. (3) Point the TFTP server of the CME phone to the UCM server running TFTP. (4) Reboot the phone (power cycle). If you have problems registering the CME phone to UCM then take into account db replication problem (operate in a PUB UCM environment by removing the SUB from the group). Restart UCM and TFTP services on PUB. Just a couple of suggestions that might fix the problem. After the phone has registered the add the CME phone MAC address into the voice register pool within the CME router. Ensure that there is no load statement within voice register global since you do not want to upload anymore firmware. Create Profile within voice register global. Fingers crossed you should be good. You can delete the CME phone entry in the UCM db. -- Vik Malhi - CCIE #13890 Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Classroom and Self-Study Cisco CCNA (RS, Voice Security), CCNP, CCVP, CCSP and CCIE (RS, Voice, Security Service Provider) Certification Training with locations throughout the United States, Europe and Australia. Be sure to check out our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com . From: Berry, Matthew J. mjbe...@krollontrack.com Date: Tue, 19 Jan 2010 21:06:38 -0600 To: OSL Group ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] SCCP to SIP Firmware Upgrade After listening to Mark Snow's comments on SCCP to SIP firmware upgrades in CUCME, I'm not surprised that I has issues tonight. I entered all the necessary commands (I think, check me on this) to allow for registration. TFTP aliases are in the config. The files are also there, preloaded for me. However, this is what I see: Jan 20 04:22:52.056: TFTP: Looking for CTLSEP001193B6EC51.tlv Jan 20 04:22:52.144: TFTP: Looking for SEP001193B6EC51.cnf.xml Jan 20 04:22:52.224: TFTP: Looking for XMLDefault.cnf.xml BR2-RTR(config-register-pool)# Jan 20 04:22:55.232: TFTP: Looking for CTLSEP0002FD3BA793.tlv Jan 20 04:22:55.292: TFTP: Looking for SEP0002FD3BA793.cnf.xml Jan 20 04:22:55.296: TFTP: Opened system:/its/XMLDefault7960.cnf.xml, fd 7, size 0 for process 339 Jan 20 04:22:55.296: TFTP: Finished system:/its/XMLDefault7960.cnf.xml, time 00:00:00 for process 339 BR2-RTR(config-register-pool)# Jan 20 04:23:05.420: TFTP: Looking for CTLSEP0002FD3BA793.tlv Jan 20 04:23:05.484: TFTP: Looking for SEP0002FD3BA793.cnf.xml Jan 20 04:23:05.484: TFTP: Opened system:/its/XMLDefault7960.cnf.xml, fd 7, size 0 for process 339 Jan 20 04:23:05.484: TFTP: Finished system:/its/XMLDefault7960.cnf.xml, time 00:00:00 for process 339 BR2-RTR(config-register-pool)# BR2-RTR#show run Jan 20 04:23:13.436: %SYS-5-CONFIG_I: Configured from console by console BR2-RTR#show Jan 20 04:23:15.612: TFTP: Looking for CTLSEP0002FD3BA793.tlv Jan 20 04:23:15.672: TFTP: Looking for SEP0002FD3BA793.cnf.xml Jan 20 04:23:15.672: TFTP: Opened system:/its/XMLDefault7960.cnf.xml, fd 7, size 0 for process 339 Jan 20 04:23:15.676: TFTP: Finished system:/its/XMLDefault7960.cnf.xml, time 00:00:00 for process 339 Does anyone have any suggestions for the CCIE Voice n00b? I am attaching a copy of my CUCME config. Any help/direction would be appreciated
[OSL | CCIE_Voice] SCCP to SIP Firmware Upgrade
After listening to Mark Snow's comments on SCCP to SIP firmware upgrades in CUCME, I'm not surprised that I has issues tonight. I entered all the necessary commands (I think, check me on this) to allow for registration. TFTP aliases are in the config. The files are also there, preloaded for me. However, this is what I see: Jan 20 04:22:52.056: TFTP: Looking for CTLSEP001193B6EC51.tlv Jan 20 04:22:52.144: TFTP: Looking for SEP001193B6EC51.cnf.xml Jan 20 04:22:52.224: TFTP: Looking for XMLDefault.cnf.xml BR2-RTR(config-register-pool)# Jan 20 04:22:55.232: TFTP: Looking for CTLSEP0002FD3BA793.tlv Jan 20 04:22:55.292: TFTP: Looking for SEP0002FD3BA793.cnf.xml Jan 20 04:22:55.296: TFTP: Opened system:/its/XMLDefault7960.cnf.xml, fd 7, size 0 for process 339 Jan 20 04:22:55.296: TFTP: Finished system:/its/XMLDefault7960.cnf.xml, time 00:00:00 for process 339 BR2-RTR(config-register-pool)# Jan 20 04:23:05.420: TFTP: Looking for CTLSEP0002FD3BA793.tlv Jan 20 04:23:05.484: TFTP: Looking for SEP0002FD3BA793.cnf.xml Jan 20 04:23:05.484: TFTP: Opened system:/its/XMLDefault7960.cnf.xml, fd 7, size 0 for process 339 Jan 20 04:23:05.484: TFTP: Finished system:/its/XMLDefault7960.cnf.xml, time 00:00:00 for process 339 BR2-RTR(config-register-pool)# BR2-RTR#show run Jan 20 04:23:13.436: %SYS-5-CONFIG_I: Configured from console by console BR2-RTR#show Jan 20 04:23:15.612: TFTP: Looking for CTLSEP0002FD3BA793.tlv Jan 20 04:23:15.672: TFTP: Looking for SEP0002FD3BA793.cnf.xml Jan 20 04:23:15.672: TFTP: Opened system:/its/XMLDefault7960.cnf.xml, fd 7, size 0 for process 339 Jan 20 04:23:15.676: TFTP: Finished system:/its/XMLDefault7960.cnf.xml, time 00:00:00 for process 339 Does anyone have any suggestions for the CCIE Voice n00b? I am attaching a copy of my CUCME config. Any help/direction would be appreciated. Digital Footprint: Skype: ciscovoiceguru cucme-sip.log Description: cucme-sip.log ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Directed Call Park- CUCM feature
Kavi, Directed call park allows you to park a call at a specific directed call park number, instead of randomly being assigned a call park number. I recreated the scenario at work, but I could not get call retrieval to fail with a status of blocked. As long as the directed call park number is in the same partition as the phone, there should not be a problem. If you are creating a separate partition for directed call park numbers, you need to make sure the calling search space assigned to your phone has the directed call park partition listed, and in the correct order. My guess is that you created another partition, but didn't add that partition to the device calling search space on your phone. That is, assuming you're implementing a line/device approach whereby the device CSS grants full site access and the line CSS provides the restrictions. Thanks, Matthew Berry Office 952 516 3748 | Mobile 952 221 2814| mjbe...@krollontrack.commailto:agutz...@krollontrack.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of kavi ten Sent: Sunday, January 17, 2010 11:16 PM To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Directed Call Park- CUCM feature Hi Guys, Did anybody manage to test this feature from CUCM Features Guide 7.0.1 Pg 4-20. If not yet can someone try share the result. Thanks, On Sat, Jan 16, 2010 at 3:33 PM, kavi ten kaucc...@gmail.commailto:kaucc...@gmail.com wrote: Hi, I have trying to practice the Directed call park(D-CP) as the step in the CUCM features guide but I can not retrive the call after its been parked. Initially I had kept the directed call park no in the null partition -It didn't work while retieving. Latter assigned a common partition assigned to the all the US phones pt-us-911 --It didn't work while retieving. Wondering why its not working I made a separate partition pt-dcp for D-CP with a separate css-dcp assisned it this D-CP, Also assigned this css-dcp exclusively to a phone -- still the same result while retrieving. Checked it in the Dna , that its Blocked. I'm choked seeing this, I have not blocked this dial-pattern any where. I assigned the Nos as follows: Number : 800 Retrived Prefix: 7 So to retrive I dial 7800, I can see the BLF-Call park on a phone configured for it. Has anybody tried this feature, can some guide me where I'm going wrong. Thanks. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Trunk Port Configuration
How do you configure speed/duplex settings on the IP phone? Is that done through CUCM? Are the steps different for CUCME? Thanks, Matthew Berry Office 952 516 3748 | Mobile 952 221 2814| mjbe...@krollontrack.commailto:agutz...@krollontrack.com From: Otto Sanchez [mailto:o...@ipexpert.com] Sent: Sunday, January 17, 2010 9:22 PM To: Arun Kumar Cc: Berry, Matthew J.; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Trunk Port Configuration Hello, If you define a specific speed/duplex setting on the port, you should do the same on the phone, you don't want to have one side auto and the other with a fixed duplex/speed configuration, On Sun, Jan 17, 2010 at 2:05 PM, Arun Kumar arunv...@gmail.commailto:arunv...@gmail.com wrote: I don't see any specific reason, as we know from lab perspective don't leave anything to default or auto-negotiate. It's better to define. On Sun, Jan 17, 2010 at 8:52 PM, Berry, Matthew J. mjbe...@krollontrack.commailto:mjbe...@krollontrack.com wrote: In workbook 1, lab 1, we are told to configure a trunk port this way: Standard Catalyst 3750 Configuration for Trunk Port Vlan 10 Name DATA State active Interface FastEthernet 1/0/2 Switchport trunk encapsulation dot1q Switchport mode trunk Switchport trunk native vlan 10 Speed 100 Duplex full Is there a specific reason that you manually set the speed and duplex instead of letting it negotiate automatically? Thanks, Matthew Berry, Sr. Unified Communications Engineer, CCVP Kroll Ontrack | 9023 Columbine Road, Eden Prairie, MN 55347 952 516 3748 | Fax 952 516 3646 | Mobile 952 221 2814| mjbe...@krollontrack.commailto:agutz...@krollontrack.com www.krollontrack.comhttp://www.krollontrack.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Phones Losing Connection with Rack
I am having an issue with phones that are connected to the Proctor Labs rack over EZVPN. If I go off-hook by lifting up the handset or pressing the speakerphone button, I will not get a dial tone. The phone is registered to CUCM. After this occurs, I will see the phone re-register. It is almost like it's not getting to Proctor Labs fast enough and it's timing out. If I call another phone and establish a call, I will also 50%+ of the time see a message on the phone display saying UCM is down. Other times, I will press End Call while a phone is ringing, and it will not respond. Has anyone had these issues? It seems that I've been having a really rough time getting basic functionality to work. Thanks, Matthew Berry, Sr. Unified Communications Engineer, CCVP Kroll Ontrack | 9023 Columbine Road, Eden Prairie, MN 55347 952 516 3748 | Fax 952 516 3646 | Mobile 952 221 2814| mjbe...@krollontrack.commailto:agutz...@krollontrack.com www.krollontrack.comhttp://www.krollontrack.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Trunk Port Configuration
In workbook 1, lab 1, we are told to configure a trunk port this way: Standard Catalyst 3750 Configuration for Trunk Port Vlan 10 Name DATA State active Interface FastEthernet 1/0/2 Switchport trunk encapsulation dot1q Switchport mode trunk Switchport trunk native vlan 10 Speed 100 Duplex full Is there a specific reason that you manually set the speed and duplex instead of letting it negotiate automatically? Thanks, Matthew Berry, Sr. Unified Communications Engineer, CCVP Kroll Ontrack | 9023 Columbine Road, Eden Prairie, MN 55347 952 516 3748 | Fax 952 516 3646 | Mobile 952 221 2814| mjbe...@krollontrack.commailto:agutz...@krollontrack.com www.krollontrack.comhttp://www.krollontrack.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Rerouting / SUBSCRIBE Calling Search Space
Can someone explain these two concepts to me? I'm not sure I understand the practical application of these options. Rerouting Calling Search Space From the drop-down list box, choose a calling search space to use for rerouting. The rerouting calling search space of the referrer gets used to find the route to the refer-to target. When the Refer fails due to the rerouting calling search space, the Refer Primitive rejects the request with the 405 Method Not Allowed message. The redirection (3xx) primitive and transfer feature also uses the rerouting calling search space to find the redirect-to or transfer-to target. SUBSCRIBE Calling Search Space Supported with the Presence feature, the SUBSCRIBE calling search space determines how Cisco Unified Communications Manager routes presence requests that come from the phone. This setting allows you to apply a calling search space separate from the call-processing search space for presence (SUBSCRIBE) requests for the phone. From the drop-down list box, choose the SUBSCRIBE calling search space to use for presence requests for the phone. All calling search spaces that you configure in Cisco Unified Communications Manager Administration display in the SUBSCRIBE Calling Search Space drop-down list box. If you do not select a different calling search space for the end user from the drop-down list, the SUBSCRIBE calling search space defaults to None. To configure a SUBSCRIBE calling search space specifically for this purpose, you configure a calling search space as you do all calling search spaces. For information on how to configure a calling search space, see the Calling Search Space Configuration section on page 50-1http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmcfg/b03csspc.html#wpxref88664 Thanks, Matthew Berry, Sr. Unified Communications Engineer, CCVP Kroll Ontrack | 9023 Columbine Road, Eden Prairie, MN 55347 952 516 3748 | Fax 952 516 3646 | Mobile 952 221 2814| mjbe...@krollontrack.commailto:agutz...@krollontrack.com www.krollontrack.comhttp://www.krollontrack.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
