Re: [OSL | CCIE_Voice] QoS - MLP LFI
Hmm, You are doing the MQC method. If you using the legacy method, then you would have define your shaping within map-class frame-relay HTH On Wed, Jul 16, 2008 at 10:24 AM, o Ninja [EMAIL PROTECTED] wrote: I have a Bootcamp workbook from Ipexpert and it has a configuration like that : class-map SIG match dscp 24 class-map RTP match dscp 46 police-map llq class RTP priority percent 33 class SIG bandwidth percent 8 class class-default fair-queue police-map Shape class class-default shape average 726800 7268 0 service-policy llq int s0/0.9 no ip address frame-relay interface-dlci 900 ppp virtual-template 1 int virtual-template 1 bandwidth 768 ip add 10.1.1.1 255.255.255.0 ppp multilink ppp multilink interleave ppp multilink fragment-delay 10 service-policy output Shape But I saw in some links the shaping being done in the FR map-class also. Date: Tue, 15 Jul 2008 19:59:13 -0500 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] QoS - MLP LFI CC: ccie_voice@onlinestudylist.com You can't do CBWFQ on a frame interface, you have to do it inside the FR map class. Jonathan On Tue, Jul 15, 2008 at 3:21 PM, o Ninja [EMAIL PROTECTED] wrote: Hello All, I have a doubt regarding to MLP. When applying this configuration QoS, which type of shaping Cisco recomends ? Shaping inside a policy-map or inside a Map-class frame-relay ? Thanks in advance ! Notícias direto do New York Times, gols do Lance, videocassetadas e muitos outros vídeos no MSN Videos! Confira já! -- Notícias direto do New York Times, gols do Lance, videocassetadas e muitos outros vídeos no MSN Videos! Confira já! http://video.msn.com/?mkt=pt-br
[OSL | CCIE_Voice] Translation Rule
Hi, Just wondering, what will the following configuration do? ! translation-rule 3 Rule 0 ^.* 902 national national Rule 1 ^.* 9002 international international Rule 2 ^.%* 92 subscriber subscriber ! Thanks.
Re: [OSL | CCIE_Voice] Nailed it down
Congrats!! very well done. On Tue, Jul 15, 2008 at 9:14 AM, ovais Iqbal [EMAIL PROTECTED] wrote: Dear All, Very glad to announce my Voice CCIE # 21482, got it today in 3rd attempt. Thanks every one for great support through out the study process, special thanks goes to Vik Malhi and ip Expert team. Once again thanks. -- Ovais Iqbal 416-294-7869
Re: [OSL | CCIE_Voice] 0 Conf max sessions
have you utilized all the dsp resources for your pri-group? On Sun, Jul 6, 2008 at 11:31 PM, Mehmet Tufekci [EMAIL PROTECTED] wrote: Hi All, I can not figure out why maximum session 0-0 is showing under conference profile. I did not enable transcoding profile yet. voice-card 0 dspfarm dsp services dspfarm ! ! ! interface Loopback0 ip address 172.3.102.1 255.255.255.255 ip ospf network point-to-point ! ! sccp local Loopback0 sccp ! dspfarm profile 1 transcode codec g711ulaw codec g729r8 shutdown ! dspfarm profile 2 conference codec g711ulaw codec g729r8 shutdown !
Re: [OSL | CCIE_Voice] callmanger wont send call to gk
Hmm, perhaps time to reboot the gatekeeper router :-) HTH On Sun, Jul 6, 2008 at 11:53 AM, [EMAIL PROTECTED] wrote: phone 1 has ccs to call 4xxx even i set the 4xxx to gk-trunk directly, it is not sending any digit out to gk. very wired.. it was working yesterday.. i start all over today, it is not working any more :-( Sara *ccievoice1 [EMAIL PROTECTED]* wrote: Well, Perhaps I will verify the route group/route list/ route pattern are configured correctly. Also, the phone's calling search space has the privilege to access to 4XXX route pattern. HTH On Sun, Jul 6, 2008 at 11:41 AM, [EMAIL PROTECTED] wrote: HQ-GW-Gatekeeper-1#sh gateke HQ-GW-Gatekeeper-1#sh gatekeeper e HQ-GW-Gatekeeper-1#sh gatekeeper endpoints GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags --- - --- - - - 172.21.1.11 1720 172.21.1.11 50985 CCM VOIP-GW H323-ID: GK-Trunk_1 Voice Capacity Max.= Avail.= Current.= 0 172.21.1.12 1720 172.21.1.12 49961 CCM VOIP-GW H323-ID: GK-Trunk_2 Voice Capacity Max.= Avail.= Current.= 0 172.21.31.2 1720 172.21.31.2 54358 CME VOIP-GW H323-ID: CME-HGK Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 3 CCM is gatekeeper controlled trunk *ccievoice1 [EMAIL PROTECTED]* wrote: do a show gatekeeper endpoint and verify if your CallManager Trunk has been registered with the gatekeeper. Also, what type of trunk you configured on the CallManager? HTH On Sun, Jul 6, 2008 at 11:21 AM, [EMAIL PROTECTED] wrote: my ccm is registered with gatekeeper, but when i configure a route pattern 4xxx to gatekeeper trunk, i dont see any digit coming in to gatekeeper. what could be wrong? Sara -- Stop! Global Warming ~ Yahoo! JAPAN Earth Projecthttp://pr.mail.yahoo.co.jp/earthproject/ -- Stop! Global Warming ~ Yahoo! JAPAN Earth Projecthttp://pr.mail.yahoo.co.jp/earthproject/ -- Stop! Global Warming ~ Yahoo! JAPAN Earth Projecthttp://pr.mail.yahoo.co.jp/earthproject/
[OSL | CCIE_Voice] Call Transfer Restriction
Hi, In CallManager Express, I can restrict call-transfer to only 4-digits internal DN# ! telephony-services transfer-system full-consult transfer-pattern 3... ! Just wondering, would I able to achieve the similar in CallManager? Thanks.
Re: [OSL | CCIE_Voice] Call Transfer Restriction
Hi, Are you referring to Block OffNet to OffNet Transfer ? But that is to restrict the transferring of an external call to an external device. But IP Phones should be considered as internal device. Thanks. On Thu, Jun 26, 2008 at 8:42 PM, Cardwell, Mark [EMAIL PROTECTED] wrote: I do believe it is a system param. Transfer offnet enabled or something like that. Cheers! Mark Cardwell | System Engineer | Presidio Networked Solutions | [EMAIL PROTECTED]| Cell: 571.225.0132 | Office: 301.623.2000| FAX: 301.313.2400 -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *ccievoice1 *Sent:* Thursday, June 26, 2008 8:26 AM *To:* OSL CCIE Voice Lab Exam *Subject:* [OSL | CCIE_Voice] Call Transfer Restriction Hi, In CallManager Express, I can restrict call-transfer to only 4-digits internal DN# ! telephony-services transfer-system full-consult transfer-pattern 3... ! Just wondering, would I able to achieve the similar in CallManager? Thanks.
Re: [OSL | CCIE_Voice] Call Transfer Restriction
Hi Chad, You were saying CFWD Calling Search Space has effect on Call Transfer? Thanks. On Fri, Jun 27, 2008 at 3:41 AM, Chad Stachowicz [EMAIL PROTECTED] wrote: ccievoice1, you can achieve this with CFWD Calling search spaces of course! Chad On 6/26/08, ccievoice1 [EMAIL PROTECTED] wrote: Hi, Are you referring to Block OffNet to OffNet Transfer ? But that is to restrict the transferring of an external call to an external device. But IP Phones should be considered as internal device. Thanks. On Thu, Jun 26, 2008 at 8:42 PM, Cardwell, Mark [EMAIL PROTECTED] wrote: I do believe it is a system param. Transfer offnet enabled or something like that. Cheers! Mark Cardwell | System Engineer | Presidio Networked Solutions | [EMAIL PROTECTED]| Cell: 571.225.0132 | Office: 301.623.2000| FAX: 301.313.2400 -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *ccievoice1 *Sent:* Thursday, June 26, 2008 8:26 AM *To:* OSL CCIE Voice Lab Exam *Subject:* [OSL | CCIE_Voice] Call Transfer Restriction Hi, In CallManager Express, I can restrict call-transfer to only 4-digits internal DN# ! telephony-services transfer-system full-consult transfer-pattern 3... ! Just wondering, would I able to achieve the similar in CallManager? Thanks.
[OSL | CCIE_Voice] CUE's General Delivery Mailbox
Hi, I know the standard way of accessing the GDM mailbox. User to access gdm message by login into their personal's mailbox first, then dial 9 to access GDM messages. Now, I am wondering can I do a not standard way of accessing the GDM mailbox, whereby a user could listen to the GDM message directly without dial 9. ??? Thanks.
[OSL | CCIE_Voice] Configuring FXS
Hi all, Wanted to configure the router as SCCP gateway and to control the FXS analog port installed in the router. I a, thinking necessary IOS commands should be as below (too bad proctorlabs vRack do not have any FXS module installed). Has anyone done that before? Please kindly advice... Thanks. ! ! stcapp ccm-group 1 stcapp ! sccp local fas0/0.210 sccp ccm 10.1.200.20 identifier 1 sccp ccm 10.1.200.21 identifier 2 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate ccm 2 priority 2 ! ! dial-peer voice 100 pots service stcapp port 0/1/0 !
Re: [OSL | CCIE_Voice] Configuring FXS
Hi, In UCM, you need to add the relevant gateway manually. So for example, add 3725 and select sccp as the protocol. Just similar to how you configure a mgcp gateway. Thanks. On Sun, Jun 22, 2008 at 2:33 AM, Stephen Collinson [EMAIL PROTECTED] wrote: Can confirm it tries to register, from sccp debug command. However could not find a device type for it to register against within UCM. AutoReg gets rejected. Registration prefix is AN Had a quick search of CCO for correct device type and it did not jump out. If you find a reference for UCM SCCP phone type let me know and I'll test it for you. HTH S -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *ccievoice1 *Sent:* 21 June 2008 18:17 *To:* OSL CCIE Voice Lab Exam *Subject:* [OSL | CCIE_Voice] Configuring FXS Hi all, Wanted to configure the router as SCCP gateway and to control the FXS analog port installed in the router. I a, thinking necessary IOS commands should be as below (too bad proctorlabs vRack do not have any FXS module installed). Has anyone done that before? Please kindly advice... Thanks. ! ! stcapp ccm-group 1 stcapp ! sccp local fas0/0.210 sccp ccm 10.1.200.20 identifier 1 sccp ccm 10.1.200.21 identifier 2 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate ccm 2 priority 2 ! ! dial-peer voice 100 pots service stcapp port 0/1/0 !
Re: [OSL | CCIE_Voice] AAR Problem!
Hi, Not sure if it is typo error. You have AAR prefix 9 with BR1 while your route pattern to BR1 is 916175222XXX IF it is not a typo, then with your AAR Prefix + BR1 External Phone Mask = 96175222002 which didn't match your BR1 Route Pattern. Br1-AAR . prefix:91 with HQ-AAR HQ-AAR . *prefix: 9* with BR1-AAR HQ phone dn [1001, 211001] BR1 phone dn [2002, 6175222002] Route patterns configured: 9.1211xxx 9.16175222xxx HTH On Mon, Jun 16, 2008 at 3:48 AM, Ahmed Hamed [EMAIL PROTECTED] wrote: Hi, I have a problem with AAR.. HQ phone is configured with location HQ-L Br1 phone is configured with location BR1-L AAR groups are configured as follows: Br1-AAR . prefix:91 with HQ-AAR HQ-AAR . prefix: 9 with BR1-AAR HQ phone dn [1001, 211001] BR1 phone dn [2002, 6175222002] Route patterns configured: 9.1211xxx 9.16175222xxx pointing to 6608 gw. Problem: Whenever I reduce location HQ-L, I try to reach destination hqphone from BR1 phone, and I get a fast busy tone.. I am not sure where the problem is.. Could anyone give me a hint? Note: Under normal conditions, I can reach hqphone from Br1 phone thru 6608 gw if I use hq phone's external mask as a destination.. Thanks, AH
Re: [OSL | CCIE_Voice] AAR Problem!
Well, I will always check and verify the calling search space and the partition. Also, I will make sure that I have the AAR Group, AAR CSS and the External Phone Mask configured on my ip phones. Lets not forget AAR Group and AAR CSS on the gateway as well. HTH On Mon, Jun 16, 2008 at 9:10 PM, Ahmed Hamed [EMAIL PROTECTED] wrote: Hi ccievoice1, Thank you for your reply. I dont's think there is a typo below, but possibly a mistake :) so... HQ-AAR should have same prefix as original route pattern e.g. 91 and 9.16175222XXX BUT I have it correct in the other way around i.e. Br1-AAR . prefix:91 with HQ-AAR and original route pattern is 9.1211xxx Any clue? AH --- On Mon, 6/16/08, ccievoice1 [EMAIL PROTECTED] wrote: From: ccievoice1 [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] AAR Problem! To: [EMAIL PROTECTED], OSL CCIE Voice Lab Exam ccie_voice@onlinestudylist.com Date: Monday, June 16, 2008, 8:20 AM Hi, Not sure if it is typo error. You have AAR prefix 9 with BR1 while your route pattern to BR1 is 916175222XXX IF it is not a typo, then with your AAR Prefix + BR1 External Phone Mask = 96175222002 which didn't match your BR1 Route Pattern. Br1-AAR . prefix:91 with HQ-AAR HQ-AAR . *prefix: 9* with BR1-AAR HQ phone dn [1001, 211001] BR1 phone dn [2002, 6175222002] Route patterns configured: 9.1211xxx 9.16175222xxx HTH On Mon, Jun 16, 2008 at 3:48 AM, Ahmed Hamed [EMAIL PROTECTED] wrote: Hi, I have a problem with AAR.. HQ phone is configured with location HQ-L Br1 phone is configured with location BR1-L AAR groups are configured as follows: Br1-AAR . prefix:91 with HQ-AAR HQ-AAR . prefix: 9 with BR1-AAR HQ phone dn [1001, 211001] BR1 phone dn [2002, 6175222002] Route patterns configured: 9.1211xxx 9.16175222xxx pointing to 6608 gw. Problem: Whenever I reduce location HQ-L, I try to reach destination hqphone from BR1 phone, and I get a fast busy tone.. I am not sure where the problem is.. Could anyone give me a hint? Note: Under normal conditions, I can reach hqphone from Br1 phone thru 6608 gw if I use hq phone's external mask as a destination.. Thanks, AH
Re: [OSL | CCIE_Voice] AAR Problem!
Hi Ahmed, Just for the sick of troubleshooting, I would just quickly create a calling search space and partition and make sure that my AAR's CSS only has visibility to the Long Distance Route Pattern (and gotto make sure is route out the local PSTN gateway). HTH On Tue, Jun 17, 2008 at 3:17 AM, Ahmed Hamed [EMAIL PROTECTED] wrote: Hi Jacob, I haven't configured any CSSs or partitions. This means anybody can call anybody. Thanks, AH --- On Mon, 6/16/08, Jacob Owen [EMAIL PROTECTED] wrote: From: Jacob Owen [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] AAR Problem! To: [EMAIL PROTECTED], OSL CCIE Voice Lab Exam ccie_voice@onlinestudylist.com Date: Monday, June 16, 2008, 6:39 PM Ahmed, Does the calling phone have the Long Distance Partition as part of it's CSS? --- Ahmed Hamed [EMAIL PROTECTED] wrote: ccievoice1, What if I don't have a CSS or a partition defined anywhere? Not even an AAR CSS! Is it a requirement to have an AAR CSS defined? Thanks, AH --- On Mon, 6/16/08, ccievoice1 [EMAIL PROTECTED] wrote: From: ccievoice1 [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] AAR Problem! To: [EMAIL PROTECTED], OSL CCIE Voice Lab Exam ccie_voice@onlinestudylist.com Date: Monday, June 16, 2008, 1:26 PM Well, I will always check and verify the calling search space and the partition. Also, I will make sure that I have the AAR Group, AAR CSS and the External Phone Mask configured on my ip phones. Lets not forget AAR Group and AAR CSS on the gateway as well. HTH On Mon, Jun 16, 2008 at 9:10 PM, Ahmed Hamed [EMAIL PROTECTED] wrote: Hi ccievoice1, Thank you for your reply. I dont's think there is a typo below, but possibly a mistake :) so... HQ-AAR should have same prefix as original route pattern e.g. 91 and 9.16175222XXX BUT I have it correct in the other way around i.e. Br1-AAR . prefix:91 with HQ-AAR and original route pattern is 9.1211xxx Any clue? AH --- On Mon, 6/16/08, ccievoice1 [EMAIL PROTECTED] wrote: From: ccievoice1 [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] AAR Problem! To: [EMAIL PROTECTED], OSL CCIE Voice Lab Exam ccie_voice@onlinestudylist.com Date: Monday, June 16, 2008, 8:20 AM Hi, Not sure if it is typo error. You have AAR prefix 9 with BR1 while your route pattern to BR1 is 916175222XXX IF it is not a typo, then with your AAR Prefix + BR1 External Phone Mask = 96175222002 which didn't match your BR1 Route Pattern. Br1-AAR . prefix:91 with HQ-AAR HQ-AAR . *prefix: 9* with BR1-AAR HQ phone dn [1001, 211001] BR1 phone dn [2002, 6175222002] Route patterns configured: 9.1211xxx 9.16175222xxx HTH On Mon, Jun 16, 2008 at 3:48 AM, Ahmed Hamed [EMAIL PROTECTED] wrote: Hi, I have a problem with AAR.. HQ phone is configured with location HQ-L Br1 phone is configured with location BR1-L AAR groups are configured as follows: Br1-AAR . prefix:91 with HQ-AAR HQ-AAR . prefix: 9 with BR1-AAR HQ phone dn [1001, 211001] BR1 phone dn [2002, 6175222002] Route patterns configured: 9.1211xxx 9.16175222xxx pointing to 6608 gw. Problem: Whenever I reduce location HQ-L, I try to reach destination hqphone from BR1 phone, and I get a fast busy tone.. I am not sure where the problem is.. Could anyone give me a hint? Note: Under normal conditions, I can reach hqphone from Br1 phone thru 6608 gw if I use hq phone's external mask as a destination.. Thanks, AH Jacob Owen CCIE #14063 (RS, Service Provider), CCVP, CCDP
[OSL | CCIE_Voice] Vol1 Section 7 Task 8 : Multicast MoH
Hi, For this Task 8, beside the configuration showed in the Proctor Guide, I am wondering are the following configuration necessary in BR1 router as well: ! ccm-manager music-on-hold ! call-manager-fallback moh music-on-hold.au multicast moh 239.1.1.3 port 16384 ! Thanks.
Re: [OSL | CCIE_Voice] VOL3 Lab 7 Task 30: VPIM
Hi Vik, Since the workbook asked to configure VPIM, can I assume that the Proctorlabs vRacks are VPIM capable? Thanks. On Wed, Jun 11, 2008 at 8:27 AM, Vik Malhi [EMAIL PROTECTED] wrote: You need to fwd messages between Unity and CUE. The steps to test this are detailed in the Proctor Guide document. Vik Malhi – CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] [EMAIL PROTECTED] Join our free online support and peer group communities: http://www.IPexpert.com/communities http://www.ipexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *ccievoice1 *Sent:* Tuesday, June 10, 2008 1:14 PM *To:* OSL CCIE Voice Lab Exam *Subject:* [OSL | CCIE_Voice] VOL3 Lab 7 Task 30: VPIM Hi, For the IPEXPERT newly released VOL3 workboook on Lab7 Task 30. We are required to create VPIM between CUE and Unity. I am just wondering, after configured the VPIM on CUE and Unity, how can I actually test the configuration? Thanks.
[OSL | CCIE_Voice] QoS Question
Hi all, I have some questions on QoS 1.) Can I define LLQ with qos percent in one class and qos bandwifth in another class? ! class-map Media match ip dscp ef ! class-map Control match ip dscp cs3 ! policy-map LLQ class media *priority percent 33* class control * bandwidth 18* class class-default fair-queue 2.) I have 1M frame-relay link between 2 sites. Based on QoS SRND, frame-relay cir = mincir = 95% of Link Speed. My question is, should I use 1M = 1000 or 1024? IF 1M = 1000, then frame-relay cir = 95 IF 2M = 1024, then frame-relay cir = 972800 Please kindly advice. Thanks
Re: [OSL | CCIE_Voice] QoS Question
Hi Mark, Thanks for the reply :-) Well, the reason for my uncertainty was because I remember some time ago, I was being told that I can't use both percent and bandwidth within a policy-map. And, now I got it clear. Thanks Mark and Devildoc for the explanation. On Thu, Jun 12, 2008 at 11:09 PM, Mark Snow [EMAIL PROTECTED] wrote: It looks like you already typed this in the router (unless you just did it in notepad) - but if you indeed did do it in the router - then you know the answer -right? :) To be more precise - yes - you can have percent on one and bandwidth on another but ONLY if they differ (percent and bandwidth) between the Strict Priority Queue and the normal Congestion Management Queues. You CANNOT have percent and bandwidth both in any of your multiple Congestion Management Queues - you must stick with whatever you choose for them all. -- Mark Snow CCIE #14073 (Voice, Security) Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: [EMAIL PROTECTED] -- Join our free online support and peer group communities: http://www.IPexpert.com/communities -- IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- On Jun 12, 2008, at 10:09 AM, ccievoice1 wrote: Hi all, I have some questions on QoS 1.) Can I define LLQ with qos percent in one class and qos bandwifth in another class? ! class-map Media match ip dscp ef ! class-map Control match ip dscp cs3 ! policy-map LLQ class media *priority percent 33* class control * bandwidth 18* class class-default fair-queue 2.) I have 1M frame-relay link between 2 sites. Based on QoS SRND, frame-relay cir = mincir = 95% of Link Speed. My question is, should I use 1M = 1000 or 1024? IF 1M = 1000, then frame-relay cir = 95 IF 2M = 1024, then frame-relay cir = 972800 Please kindly advice. Thanks
Re: [OSL | CCIE_Voice] Upgrade to Worksbooks
As from what I understand, If you got those workbooks from IP Expert, then they will provide the updated workbooks in pdf format. Just login to your IP Expert account. Hope I am right. On Fri, Jun 6, 2008 at 3:37 PM, Ashraf Hannoush [EMAIL PROTECTED] wrote: Hi, I have: IPexpert's Ultimate Lab Preparation Workbook v4.0 and IPexpert's CCIE Voice Proctor Guide v4.0 Is it possible to get a free upgrade on these products? Please advise, Ashraf
Re: [OSL | CCIE_Voice] Extension Mobility on IP Blue
I just configured as usual Cisco IP Phone and it worked. So, what help do you need exactly? HTH On Fri, May 30, 2008 at 12:15 AM, Ahmed Hamed [EMAIL PROTECTED] wrote: Hi, Any idea how to implement Extension Mobility on IP Blue? I am trying to configure SERVICES button in the IP Blue but with no luck! Please advise, AH
Re: [OSL | CCIE_Voice] http://www.proctorlabs.com/ link down?
I had this problem several times as well!! Ever since proctorlabs implemented the auto grading system, their website has been slow and not responsive :-( On Sun, May 25, 2008 at 2:10 PM, [EMAIL PROTECTED] wrote: i am having a session now, and i couldnt access the link now -- GANBARE! NIPPON! Win your ticket to Olympic Games 2008.http://pr.mail.yahoo.co.jp/ganbare-nippon/
Re: [OSL | CCIE_Voice] BACD drop-through mode
param drop-through-prompt .au is an optional param. Not required for bacd drop through to be working HTH On Tue, May 20, 2008 at 12:31 PM, Mike O [EMAIL PROTECTED] wrote: Looks like you are missing... param drop-through-prompt .au -Mike CCIE #17982 - Original Message - *From:* Gregory Jost (grjost) [EMAIL PROTECTED] *To:* ccievoice ccie_voice@onlinestudylist.com *Sent:* Monday, May 19, 2008 10:30 PM *Subject:* [OSL | CCIE_Voice] BACD drop-through mode I'm having trouble getting drop-through mode to work. The aa works fine. Can anyone spot my problem? It is as if a parameter is incorrect (e.g. immediately disconnects). application service queue flash:app-b-acd-2.1.0.0.tcl param aa-hunt3 3501 param queue-len 15 param aa-hunt1 3002 param queue-manager-debugs 1 param aa-hunt2 3003 param number-of-hunt-grps 3 ! service dt flash:app-b-acd-aa-2.1.0.0.tcl paramspace english index 0 param handoff-string dt paramspace english language en paramspace english location flash: param service-name queue param aa-pilot 3400 param number-of-hunt-grps 1 param drop-through-option 3 ! service aa flash:app-b-acd-aa-2.1.0.0.tcl paramspace english index 1 param number-of-hunt-grps 2 param handoff-string aa param dial-by-extension-option 5 paramspace english language en param max-time-vm-retry 2 param aa-pilot 3500 paramspace english location flash: param second-greeting-time 60 param welcome-prompt _bacd_welcome.au param call-retry-timer 15 param max-time-call-retry 700 param voice-mail 3600 param service-name queue ! dial-peer voice 3500 pots service aa incoming called-number 3500 direct-inward-dial port 0/0/0:15 ! dial-peer voice 3501 voip service aa destination-pattern 3500 session target ipv4:172.1.102.1 incoming called-number 3500 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 3400 pots service dt incoming called-number 3400 direct-inward-dial port 0/0/0:15 ! dial-peer voice 3401 voip service dt destination-pattern 3400 session target ipv4:172.1.102.1 incoming called-number 3400 dtmf-relay h245-alphanumeric codec g711ulaw no vad Greg Jost Network Consulting Engineer Unified Communications Practice Cisco Systems, Inc. 214-274-1922
Re: [OSL | CCIE_Voice] Compressed RTP
Because, Compress Header IP RTP only supported on OUTBOUND interface. HTH On Sun, May 18, 2008 at 2:55 PM, Paul and Bobs [EMAIL PROTECTED] wrote: Does anyone know why I wuld be getting the following error when applying policy-map inbound BR1-RTR(config-subif)#service-policy input CRTP Header compression: Can be enabled as an output feature only On Sat, May 17, 2008 at 9:02 PM, Alex [EMAIL PROTECTED] wrote: Hi there, You have to apply service-policy with cRTP on both outbound/egress and inbound/ingress interfaces at the same time. Rgds Alex - Original Message - *From:* Paul and Bobs [EMAIL PROTECTED] *To:* ccie_voice@onlinestudylist.com *Sent:* Saturday, May 17, 2008 2:35 AM *Subject:* [OSL | CCIE_Voice] Compressed RTP I am running the following config on my lab for compressed RTP. when I apply it to the serial interface outbound I loose the audio stream in that direction. call remains up and one way audio is there. When I remove the service policy output the audio come back. Any ideas Thanks class-map match-any RTP match dscp ef match access-group 100 class-map match-any SIG match dscp cs3 match dscp af31 ! ! policy-map CRTP class RTP compress header ip rtp access-list 100 permit udp 10.61.113.0 0.0.0.255 10.61.111.0 0.0.0.255range 16384 32767 access-list 100 permit udp 10.61.113.0 0.0.0.255 range 16384 32767 10.61.111.0 0.0.0.255
Re: [OSL | CCIE_Voice] Voice Translation Rule
Translation profile allowed you to bind multiple translation-rule within. translation-profile CISCO translate called 1 translate calling 2 voice-port 0/2/0:23 translation-profile incoming 110 HTH On 5/19/08, WorkerBee [EMAIL PROTECTED] wrote: What is the difference between the following? Can I say that 'translate' command only strictly applied to voice-port inbound only? Whereas translation-profile is more flexible (inbound/outbound)? In the example below, is there a priority of which command to invoke? translation-rule 110 Rule 0 ^222 21 translation-rule 1 Rule 0 ^222 21 voice-port 0/2/0:23 translation-profile incoming 110 translate called 1 = CCO == translation-profile (voice-port) - Use the translation-profile command to assign a predefined translation profile to a voice port. translate (voice-port) - To apply a translation rule to manipulate dialed digits on an inbound POTS call leg, use the translate command in voice-port configuration mode.
[OSL | CCIE_Voice] RmCm Subsystem Initiliazing
Hi all, Anyone know how to troubleshoot RmCm Subsystem? It keep in Initializing state, reboot of servers not helping at all. Please kindly advice. Thanks.
[OSL | CCIE_Voice] MGCP Gateway Issue
Hi all, I am having problem getting my mgcp gateway to register to CallManager. I tried reset the gateway from CallManager, and issue no mgcp/mgcp in the router. And, end up rebooting the CallManager and the router. But still no luck :-( Please kindly advice. Here are some outputs: P21-BR1-RTR# *May 15 01:01:58.203: cmapp_mgr_process_ev_active_host_failed: Active host 0 (10.21.200.20) failed *May 15 01:01:58.203: cmbh_tcp_open_ind: TCP open failed for 10.21.200.21, calling callback. *May 15 01:01:58.207: cmbh_tcp_open_ind: TCP open failed for 10.21.200.20, calling callback. *May 15 01:01:58.207: cmapp_mgr_process_ev_active_host_failed: Active host 0 (10.21.200.20) failed P21-BR1-RTR# P21-BR1-RTR#s *May 15 01:02:13.203: cmapp_mgr_process_ev_active_host_failed: Active host 0 (10.21.200.20) failed *May 15 01:02:13.203: cmbh_tcp_open_ind: TCP open failed for 10.21.200.20, calling callback. *May 15 01:02:13.203: cmbh_tcp_open_ind: TCP open failed for 10.21.200.21, calling callback. *May 15 01:02:13.207: cmapp_mgr_process_ev_active_host_failed: Active host 0 (10.21.200.20) failed P21-BR1-RTR#sh ccm MGCP Domain Name: P21-BR1-RTR PriorityStatus Host Primary Down 10.21.200.20 First BackupDown 10.21.200.21 Second Backup None Current active Call Manager:None Backhaul/Redundant link port: 2428 Failover Interval: 30 seconds Keepalive Interval: 15 seconds Last keepalive sent:00:47:17 UTC May 15 2008 (elapsed time: 00:14:58) Last MGCP traffic time: 01:00:13 UTC May 15 2008 (elapsed time: 00:02:02) Last failover time: None Last switchback time: None Switchback mode:Graceful MGCP Fallback mode: Enabled/ON Last MGCP Fallback start time: 00:48:06 UTC May 15 2008 Last MGCP Fallback end time:None MGCP Download Tones:Disabled Backhaul/Redundant link is down Configuration Error History: FAX mode: cisco P21-BR1-RTR#sh mgcp MGCP Admin State ACTIVE, Oper State ACTIVE - Cause Code NONE MGCP call-agent: 10.21.200.20 Initial protocol service is MGCP 0.1
Re: [OSL | CCIE_Voice] Dial peer preference timers.
Is that a VOIP dial-peer? If it is, then you can use this: voice class h323 1 h225 timeout tcp establish 3 and apply to you dial-peer dial-peer voice 10 voip destination-pattern 1234 session target ipv4:10.10.10.1 voice-class h323 1 dial-peer voice 11 voip destination-pattern 1234 session target ipv4:10.10.10.2 voice-class h323 1 preference 1 HTH On Sat, May 10, 2008 at 10:48 AM, Paul and Bobs [EMAIL PROTECTED] wrote: HI I am looking for the timer on the preference for the dial peers. If one dial peer doesnt work ( for whatever reason - say session target is unavailable) the second preference takes a while to take over. I am looking for this timer. Paul
Re: [OSL | CCIE_Voice] CCM : location based CAC MOH bandwidthconsideration
Well, If you read the MOH topic in CCM 4.x SRND, Multicast MOH won't contribute to your Location bandwidth. However, Unicast MOH does. On Wed, Apr 30, 2008 at 10:01 PM, FrogOnDSCP46EF [EMAIL PROTECTED] wrote: Christian, I think one stream (MOH) should consume what is configured in the srevice parameter under mediaapp. The case I depicted here is G729 region (hq to siteB). So... Back to the my original question, bandwidth consideration, 2xg729 calls vs 3 x G729 calls. If you set 3 calls, (considering 1xg729 for MOH), you may get 0 in HA section. Frog On Wed, Apr 30, 2008 at 11:32 PM, Christian Narvaez [EMAIL PROTECTED] wrote: You can test it using Permon , and selecting the Performance Object Cisco Locations then see how the BandwidthAvailable varies. -Original Message- From: [EMAIL PROTECTED] on behalf of FrogOnDSCP46EF Sent: Wed 4/30/2008 5:45 AM To: CCIE Voice Maillist Subject: [OSL | CCIE_Voice] CCM : location based CAC MOH bandwidthconsideration Folks, Allow 2 calls between HQ and SiteB , HQ to siteb are in G729 region. 2 calls = 24kbps per call x 2 = 48 kbps What about MOH bandwidth if HQ is pumping M.Cast MOH to SITEB? Think about when both calls are occupied and 48kbps bandwidth CAC is exhausted and we also want to put someone on MOH? How CCM maths work in that situation? Frog -- Smile, you'll save someone else's day! Frog -- Smile, you'll save someone else's day! Frog
Re: [OSL | CCIE_Voice] CCM : location based CAC MOH bandwidthconsideration
CCM 4.x SRND, Music on Hold -- Call Admission Control and MoH Cisco Unified CallManager locations-based call admission control is capable of tracking unicast MoH streams traversing the WAN but not multicast MoH streams HTH On Thu, May 1, 2008 at 10:44 AM, ccievoice1 [EMAIL PROTECTED] wrote: Well, If you read the MOH topic in CCM 4.x SRND, Multicast MOH won't contribute to your Location bandwidth. However, Unicast MOH does. On Wed, Apr 30, 2008 at 10:01 PM, FrogOnDSCP46EF [EMAIL PROTECTED] wrote: Christian, I think one stream (MOH) should consume what is configured in the srevice parameter under mediaapp. The case I depicted here is G729 region (hq to siteB). So... Back to the my original question, bandwidth consideration, 2xg729 calls vs 3 x G729 calls. If you set 3 calls, (considering 1xg729 for MOH), you may get 0 in HA section. Frog On Wed, Apr 30, 2008 at 11:32 PM, Christian Narvaez [EMAIL PROTECTED] wrote: You can test it using Permon , and selecting the Performance Object Cisco Locations then see how the BandwidthAvailable varies. -Original Message- From: [EMAIL PROTECTED] on behalf of FrogOnDSCP46EF Sent: Wed 4/30/2008 5:45 AM To: CCIE Voice Maillist Subject: [OSL | CCIE_Voice] CCM : location based CAC MOH bandwidthconsideration Folks, Allow 2 calls between HQ and SiteB , HQ to siteb are in G729 region. 2 calls = 24kbps per call x 2 = 48 kbps What about MOH bandwidth if HQ is pumping M.Cast MOH to SITEB? Think about when both calls are occupied and 48kbps bandwidth CAC is exhausted and we also want to put someone on MOH? How CCM maths work in that situation? Frog -- Smile, you'll save someone else's day! Frog -- Smile, you'll save someone else's day! Frog
Re: [OSL | CCIE_Voice] 6500 QoS - Policer configuration
Kindly noticed the kbits used by the BURST and Kbits you have mentioned. They are different ... On Tue, Apr 29, 2008 at 12:46 AM, Christian Narvaez [EMAIL PROTECTED] wrote: Well, that is the source of the confusion, because if I configure the police as set qos policer aggregate SIG rate 32 burst 8000 policed-dscp , then the show qos policer runtime all shows the busts as 7936 Kb ( that means near 8 Megabits !) Aggregate name Avg. rate (kbps) Burst size (kbits) Normal action --- -- - SIG 327936 policed-dscp Excess rate (kbps) Excess burst size (kbits) Excess action -- - - 31457280 31744 policed-dscp ACL attached CCM And following the recommendation Bust = Rate / 4000 = 32 Kbits / 4 = 8 Kbits Then I think it should be set qos policer aggregate SIG rate 32 burst 8 policed-dscp But I am reading from several sources which configure this in different ways. Anybody have any clarification about this ? -Original Message- From: [EMAIL PROTECTED] on behalf of ccievoice1 Sent: Mon 4/28/2008 12:32 PM To: Christian Narvaez Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] 6500 QoS - Policer configuration As per SRND, should be rate 32 burst 8000 On Tue, Apr 29, 2008 at 12:19 AM, Christian Narvaez [EMAIL PROTECTED] wrote: I have a confusion. In a 6500, if I want to configure a policer of 32 Kb rate and 8 Kb of burst. Which of these three sentences is correct ? a) set qos policer aggregate SIG rate 32000 burst 8000 policed-dscp or b) set qos policer aggregate SIG rate 32 burst 8 policed-dscp or c) set qos policer aggregate SIG rate 32 burst 8000 policed-dscp
Re: [OSL | CCIE_Voice] Record a Greeting for a Call Handler
Why not? Just make sure your softphone is registered to the CallManager. I have been doing that for 30++ sessions On Mon, Apr 28, 2008 at 10:53 AM, Ahmed Hamed [EMAIL PROTECTED] wrote: Hi, I am trying to record a greeting for a Call Handler (non subscriber). How is it possible if I am connecting in a Virtual session to Proctor Lab. I am not using any hardware ip phones, but just ip blue/ ip communicator through Cisco vpn client. Thanks, AH -- Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.http://us.rd.yahoo.com/evt=51733/*http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ
Re: [OSL | CCIE_Voice] Record a Greeting for a Call Handler
Yes, your phone should ring :-) And you will hear a beep and you will start recording your own voice. On Mon, Apr 28, 2008 at 12:30 PM, Ahmed Hamed [EMAIL PROTECTED] wrote: Hi, Softphone is registered to CallManager but, when I enter Extension details and IP address of Unity server in Media Master Options, I get a message after some time saying something like phone did not answer. I should get a ring on the softphone, right? Any clue? Thanks AH *ccievoice1 [EMAIL PROTECTED]* wrote: Date: Mon, 28 Apr 2008 11:03:38 +0800 From: ccievoice1 [EMAIL PROTECTED] To: Ahmed Hamed [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] Record a Greeting for a Call Handler CC: ccievoice ccie_voice@onlinestudylist.com Why not? Just make sure your softphone is registered to the CallManager. I have been doing that for 30++ sessions On Mon, Apr 28, 2008 at 10:53 AM, Ahmed Hamed [EMAIL PROTECTED] wrote: Hi, I am trying to record a greeting for a Call Handler (non subscriber). How is it possible if I am connecting in a Virtual session to Proctor Lab. I am not using any hardware ip phones, but just ip blue/ ip communicator through Cisco vpn client. Thanks, AH -- Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.http://us.rd.yahoo.com/evt=51733/*http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ
Re: [OSL | CCIE_Voice] how should the phone ports be configured on theES-SW module..
So, does ATA186/188 have the same config apply? Thanks. On Sun, Apr 27, 2008 at 6:43 AM, Christian Narvaez [EMAIL PROTECTED] wrote: For an ESW Port with IPPhone+PC, It should be something like this: inter fastethernet 1/0 switchport trunk encapsulation dot1q switchport mode trunk switchport trunk native vlan 100 Data Vlan switchport voice vlan 200 Voice Vlan -Original Message- From: [EMAIL PROTECTED] on behalf of Suresh Velayudhan Sent: Sat 4/26/2008 5:59 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] how should the phone ports be configured on theES-SW module.. I am trying to configure the HWIC-9-ESSW module for connecting to the IP phones. For normal switch ports this should be configured as an access port. I am can get the phones to register when configured as an access port, it works fine when configured as a trunk port. Want to make sure if trunk is the right configuration. Thanks Suresh
[OSL | CCIE_Voice] FXS voice port on Proctorlabs Voice vRack
Hi, Not sure if Proctorlabs/IP Expert is planning to have some FXS/FXO voice port installed for their vRack? Though FXS/FXO might just involved simple configuration, but looking at which, FXS and FXO are included in the CCIE Voice blueprint. Definitely would be good to have real hands-on practice. Thanks.
Re: [OSL | CCIE_Voice] 6500 marking
Hi Vik Malhi, Does that mean, if just to mark the signaling traffic in CallManager and ignore the Media traffic, then only fields would be in CallManager Enterprise Parameters (which default is configured as CS3)? Thanks. On Fri, Apr 18, 2008 at 7:44 AM, Vik Malhi [EMAIL PROTECTED] wrote: If your told to mark signaling and media traffic from CallManager then you are absolutely correct- the IP Voice Media Streaming App will produce media packets and there is a service parameter to change the markings if you want to trust them. If you are told to just ensure the signaling traffic is marked correctly then you can ignore the media traffic emerging from the IP Voice Media Streaming service. Vik Malhi – CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] [EMAIL PROTECTED] Join our free online support and peer group communities: http://www.IPexpert.com/communities http://www.ipexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Gregory Jost (grjost) *Sent:* Thursday, April 17, 2008 12:15 PM *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] 6500 marking I don't see it mentioned anywhere that CRS, Attendant Console Queue, MOH, Annunciator, MTP, etc. all use RTP. If you're not allowed to trust DSCP from CCM/CRS and Unity, then these too should be marked on the server ports, not just signaling protocols. Thoughts? Greg Jost Network Consulting Engineer Unified Communications Practice Cisco Systems, Inc. 214-274-1922
Re: [OSL | CCIE_Voice] B-ACD namespace error
Well, you can just ignore the error message. It appeared every time you entered a param syntax. HTH On Sun, Apr 13, 2008 at 10:02 AM, Jacob Owen [EMAIL PROTECTED] wrote: Sara, I get that same error when I configure CME Scripting, but I will say that CME Scripting is one of my very weak areas so hopefully someone else will chime in and let us both know if this is an error to be concerned with or not. --- [EMAIL PROTECTED] wrote: i am testing the B-acd feature of cme when i enter the example config from lab14 on my own router, there are a few namespace error, what should i do, can anyone help? ccme-cue(config-app-param)# param handoff-string aa Warning: parameter handoff-string has not been registered under aa namespace thanks in advance Sara - GANBARE! NIPPON! Win your ticket to Olympic Games 2008. Jacob Owen CCIE #14063 (RS, Service Provider), CCVP, CCDP __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
Re: [OSL | CCIE_Voice] B-ACD namespace error
Oops, my bad. Sorry Chad to get you wrong. Anyway, I never needed to reload the TCL script for the param to work. I mean, in my lab it just worked after entering the necessary BACD syntax. Thanks. On Sun, Apr 13, 2008 at 10:26 AM, Chad Stachowicz [EMAIL PROTECTED] wrote: ccievoice1, What I meant to say is anytime you have B-ACD loaded and you type a change int he params, and it sasy it isn't registered under the aa namespace. You need to reload the application in order to use the new settings. CHad On 4/12/08, ccievoice1 [EMAIL PROTECTED] wrote: New params ? interesting... So what is the new param for param handoff-string param second-greeting-time param voice-mail and rest of the syntax started with param ? Please advice, thanks. On Sun, Apr 13, 2008 at 10:15 AM, Chad Stachowicz [EMAIL PROTECTED] wrote: Yeah but what that should tell you is to reload the tcl to use the new param's!!! Chad On 4/12/08, ccievoice1 [EMAIL PROTECTED] wrote: Well, you can just ignore the error message. It appeared every time you entered a param syntax. HTH On Sun, Apr 13, 2008 at 10:02 AM, Jacob Owen [EMAIL PROTECTED] wrote: Sara, I get that same error when I configure CME Scripting, but I will say that CME Scripting is one of my very weak areas so hopefully someone else will chime in and let us both know if this is an error to be concerned with or not. --- [EMAIL PROTECTED] wrote: i am testing the B-acd feature of cme when i enter the example config from lab14 on my own router, there are a few namespace error, what should i do, can anyone help? ccme-cue(config-app-param)# param handoff-string aa Warning: parameter handoff-string has not been registered under aa namespace thanks in advance Sara - GANBARE! NIPPON! Win your ticket to Olympic Games 2008. Jacob Owen CCIE #14063 (RS, Service Provider), CCVP, CCDP __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
Re: [OSL | CCIE_Voice] VG248 port 0
Should be used for voicemail. On Fri, Apr 11, 2008 at 9:26 AM, Paul and Bobs [EMAIL PROTECTED] wrote: What is the first port on VG248 used for. Port 0??
Re: [OSL | CCIE_Voice] B-acd script in Dynamips
Yes, BACD is able to work in dynamips. HTH On Wed, Apr 9, 2008 at 3:40 PM, Balamurugan Singaram [EMAIL PROTECTED] wrote: Hi, When I try b-acd script in Dynamips, I getting the following error message %CALL_CONTROL-6-APP_NOT_FOUND:. Could please let me know can we run b-acd scripts in Dynamips, it will work or I am missing. Thanks, Bala Send instant messages to your online friends http://uk.messenger.yahoo.com
Re: [OSL | CCIE_Voice] B-acd script in Dynamips
Definitely you need the tcl files to be located in your router flash: You can use tftp to upload the tcl files. On Wed, Apr 9, 2008 at 4:28 PM, Balamurugan Singaram [EMAIL PROTECTED] wrote: no I am not able to tcl files in flash, but I have copy all the tcl files in dynamips image folder, could please let me know how to upload tcl files in flash [dynamips] --- PSTN#sh flash System CompactFlash directory: File Length Name/status 1 187715 crashinfo_20020301-012431 [16777212 bytes used, 0 available, 16777212 total] 16384K bytes of ATA System CompactFlash (Read/Write) PSTN#dir flash: Directory of flash:/ 1 -rw- 187715no date crashinfo_20020301-012431 16777212 bytes total (0 bytes free) -- *ccievoice1 [EMAIL PROTECTED]* wrote: When you do dir flash: Can you see all the tcl files in the flash: ?? HTH On Wed, Apr 9, 2008 at 4:00 PM, Balamurugan Singaram [EMAIL PROTECTED] wrote: Hi, I have upload the b-acd script in Image folder in dynamips, and following is my config, could please let me know what I am missing: voice service voip allow-connections h323 to h323 allow-connections h323 to sip no supplementary-service h450.2 no supplementary-service h450.3 ephone-hunt 1 longest-idle pilot list 4001, 4002 timeout 10 ephone-hunt 2 longest-idle pilot list 4101, 4102 timeout 10 application service queue flash:app-b-acd-2.1.0.0.tcl param number-of-hunt-grps 2 param aa-hunt2 param aa-hunt3 param queue-len 15 param queue-manager-debugs 1 ! service aa flash:app-b-acd-aa-2.1.0.0.tcl paramspace english index 1 paramspace english language en paramspace english location flash: param service-name queue param handoff-string aa param aa-pilot 8005550123 param welcome-prompt _bacd_welcome.au param number-of-hunt-grps 2 param dial-by-extension-option 1 param second-greeting-time 60 param call-retry-timer 15 param max-time-call-retry 700 param max-time-vm-retry 2 param voice-mail 5003 ! dial-peer voice 222 voip service aa destination-pattern 8005550123 session target ipv4:192.168.1.1 incoming called-number 8005550123 dtmf-relay h245-alphanumeric codec g711ulaw no vad int loopback0 ip address 192.168.1.1 Thanks, *ccievoice1 [EMAIL PROTECTED]* wrote: Yes, BACD is able to work in dynamips. HTH On Wed, Apr 9, 2008 at 3:40 PM, Balamurugan Singaram [EMAIL PROTECTED] wrote: Hi, When I try b-acd script in Dynamips, I getting the following error message %CALL_CONTROL-6-APP_NOT_FOUND:. Could please let me know can we run b-acd scripts in Dynamips, it will work or I am missing. Thanks, Bala Send instant messages to your online friends http://uk.messenger.yahoo.com Send instant messages to your online friends http://uk.messenger.yahoo.com Send instant messages to your online friends http://uk.messenger.yahoo.com
Re: [OSL | CCIE_Voice] dial-peer patterns
Try adding a T at the end of your local dial-peer ? HTH On Wed, Apr 9, 2008 at 11:09 PM, jason sung [EMAIL PROTECTED] wrote: Has anyone been able to get long distance working using the following two dial-peer patterns 1.Local 9[1-9].. 2. LD 91[1-9]..[1-9].. I tweaked it all the possible ways but every time I dialed the LD number the router snapped it and sent it out as local. I do know that both patterns are possible matches when you start dialing but shouldn't the router wait for rest of the digits to come in and than make a decision? I know this is most likely not going to happen in lab but I just wanted to get this working. It is bugging me.
Re: [OSL | CCIE_Voice] First attempt at IPEXPERT lab
Do you configured the trunk native vlan on both Cat6 and HQ-RTR? On Thu, Apr 10, 2008 at 6:19 AM, Paul and Bobs [EMAIL PROTECTED] wrote: Hi guys Not sure if i am missing something here. I have basic config applied. I am in POD 12 and have 2/7-9 = IP phones and ATA 2/10 HQ-RTR 2/11 = CCM (Pub,Sub and Unity) 6608 set port enable 2/7-11 set vlan 120 2/7-9 set vlan 220 2/10-11 set port auxiliaryvlan 2/7-9 120 set trunk 2/10 on dot1q 120,220 set trunk 2/11 off dot1q 1-4094 HQ-RTR interface f0/0 no shut interface fa0/0.220 encapsulation dot1q 220 ip address 10.2.200.3 255.255.255.0 no shut I can ping from BR1-RTR to 10.2.200.3 but not to my Call Managers. Not sure if I am missing something on the CATOS. Paul
[OSL | CCIE_Voice] Not able to access proctorlabs.com, again?!
Hi all, anyone having difficulty accessing the proctorlabs right now? Thanks.
Re: [OSL | CCIE_Voice] 6608 T1 registration
You are just registering the T1 PRI port on 6608. You need to add the xcode and confb as well. FOR Example: Xcode set port voice interface 4/5 dhcp enable vlan 220 Confb set port voice interface 4/6 dhcp enable vlan 220 Then, get the MAC Address for 4/5 and 4/6 and add them to CCM Xcode and CCM Confb respectively. HTH On Thu, Apr 10, 2008 at 9:01 AM, Paul and Bobs [EMAIL PROTECTED] wrote: HI Is there anything else that need to be done to register the 6608 gateway into callmanager. I have set port voice interface 4/4 dhcp enable vlan 220 show port 4/4 get mac add 6608 T1 (PRI) use mac reset. I think im missing a step as the conference and xcode are not registering either. This is first attempt on IPEXPERT lab so hence first attempt with 6608 resources. Loads of fun. Paul
Re: [OSL | CCIE_Voice] IP Expert Lab 6608 gateway
Hmm, I never encountered any issue during my vRack session ... On Thu, Apr 10, 2008 at 10:45 AM, Paul and Bobs [EMAIL PROTECTED] wrote: HI all If anyone has had much luck with reistering the 6608 gateway to CCM in the proctorlabs I would really like to hear from you. Interested in what steps you took to get it registered. Thanks
Re: [OSL | CCIE_Voice] MGCP and SRST
If the WAN link is down, then shouldn't affect the AAR call as AAR is utilizing the PSTN... and the phones should able to preserve the call HTH On Tue, Apr 8, 2008 at 10:17 PM, FrogOnDSCP46EF [EMAIL PROTECTED] wrote: I was just wondering how this tracking business will be done in SRST. Assume AAR connection is up between siteB HQ and suddenly the serial link goes down. Frog --- I think somehow the status of the serial interface is linked to the PRI link by some connection ID or something because when the serial interface is down, that connection ID is lost. When you bring the serial interface backup again, the serial interface must generate a new ID or something and it cannot associate that to the PRI link.
Re: [OSL | CCIE_Voice] MGCP and H323 Gateways
For MGCP with PRI backhaul to CallManager, service mgcpapp is not required. HTH On Wed, Apr 9, 2008 at 10:53 AM, Jonathan Charles [EMAIL PROTECTED] wrote: On the MGCP router add this dial-peer voice 3000 pots port 0/1/0:15 service mgcpapp That is the only obvious thing missing... Jonathan On Tue, Apr 8, 2008 at 9:47 PM, Paul and Bobs [EMAIL PROTECTED] wrote: HI Guys Got a teaser that bugging me now. I have in my lab HQ and Br1 connected with E1 crossover to simulate PSTN as best I can. I have Br1 connected to CCM with MGCP and HQ configured as H323 gateway in CCM. I have respective route paterns for the remote sites configured to point to the relevent gateways. When I try to make a call either way i can see the isdn q931 messages coming in on the HQ router and the call is answered (in otherwords, I get a dialtone on the Br1 Ph1 when trying to call HQ Ph1). I have attahced the configs below for some assistance. *** *BR1 CONFIG *** service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname BR1-RTR ! boot-start-marker boot system flash:c2801-adventerprisek9_ivs-mz.124-15.T3.bin boot-end-marker ! logging buffered 51200 warnings ! no aaa new-model clock timezone AEST 10 clock summer-time AEDT recurring last Sun Oct 2:00 last Sun Mar 2:00 network-clock-participate wic 1 ip cef ! ! ! ! ip domain name iptlab.local ip auth-proxy max-nodata-conns 3 ip admission max-nodata-conns 3 ! multilink bundle-name authenticated ! isdn switch-type primary-qsig ! voice-card 0 dsp services dspfarm ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! crypto pki trustpoint TP-self-signed-3566742966 enrollment selfsigned subject-name cn=IOS-Self-Signed-Certificate-3566742966 revocation-check none rsakeypair TP-self-signed-3566742966 ! ! crypto pki certificate chain TP-self-signed-3566742966 certificate self-signed 01 3082023F 308201A8 A0030201 02020101 300D0609 2A864886 F70D0101 04050030 31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274 69666963 6174652D 33353636 37343239 3636301E 170D3038 30343037 32333239 33325A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649 4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D33 35363637 34323936 3630819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281 8100DFA5 C8BF2A0D 6FF5F6F4 7D50FE3D 44386FAD 7884AC3D 845C472D A70AD441 7646F9A4 B92AC281 D1FD75F4 20AE3963 01AA0B20 98CD7801 339CBB46 D55A9B88 7EF00720 5384C2E5 C197C70E 11BDE619 796E4C3D 842C5CD7 8744A436 6BEC79A1 1B1B7603 2F97C7A7 B4785F92 FA4C054C 550FCCE8 7E5F5B79 32D6E0B8 56F33AA9 9DF50203 010001A3 67306530 0F060355 1D130101 FF040530 030101FF 30120603 551D1104 0B300982 07425231 2D525452 301F0603 551D2304 18301680 14E18894 7330012A 686D5557 5893B881 DD72662F 85301D06 03551D0E 04160414 E1889473 30012A68 6D555758 93B881DD 72662F85 300D0609 2A864886 F70D0101 04050003 8181008C 2F39C296 171E30CA 098B3D7E 0341861F 0EDBAC13 AF3828E3 8EB55990 ADB35967 38D0D7CD 7FBB6B9C 0210F1B0 952C20DD A15718FC 96F48A7C B1454A6E A67290E1 7AB205EF F3FAAC14 27719656 A7BE1162 2A343C28 B6C953B0 26FFD202 416DA3EB CA29F1AE 6EA3F731 62BF4FB8 062237AF 3F68A61C 217D98CF 911AD392 62B88E quit ! ! ! ! controller E1 0/1/0 pri-group timeslots 1-3,16 service mgcp ! ! class-map match-any RTP match dscp ef match access-group 101 class-map match-any SIG match dscp af31 match dscp cs3 match access-group 102 ! ! policy-map LLQ class RTP priority percent 33 set dscp ef class SIG bandwidth 8 set dscp cs3 class class-default fair-queue set dscp default ! ! ! ! ! interface Loopback0 ip address 10.61.127.1 255.255.255.255 ! interface FastEthernet0/0 description BR1 LAN no ip address duplex auto speed auto ! interface FastEthernet0/0.112 encapsulation dot1Q 112 ip address 10.61.112.1 255.255.255.0 ip helper-address 10.61.111.4 ip pim sparse-dense-mode ! interface FastEthernet0/0.113 encapsulation dot1Q 113 ip address 10.61.113.1 255.255.255.0 ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! interface Serial0/1/0:15 no ip address encapsulation hdlc isdn switch-type primary-qsig isdn incoming-voice voice isdn bind-l3 ccm-manager isdn outgoing display-ie isdn outgoing ie redirecting-number no cdp enable ! interface Serial0/2/0 no ip address encapsulation frame-relay no fair-queue frame-relay traffic-shaping
Re: [OSL | CCIE_Voice] Dialing HQ phones to CME BACD-AA number
Is your prompt param configured correctly? Since you said it is connected and hear silence ... On Mon, Apr 7, 2008 at 8:13 PM, FrogOnDSCP46EF [EMAIL PROTECTED] wrote: Hi Johathan, Thats already enabled. Xcoder is also inplace. Still no go... Call shows its connected on the IP phone screen but silence and after 30 second it gets disconnected. THe debug voip dialpeers doesn't tell anything. On Mon, Apr 7, 2008 at 12:56 PM, Jonathan Charles [EMAIL PROTECTED] wrote: Enable an IPIPGW voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip By default all of the above are disabled. Jonathan On Sun, Apr 6, 2008 at 7:32 PM, FrogOnDSCP46EF [EMAIL PROTECTED] wrote: Anybody has working solution for this scenario? HQ phones to CME BACD AA won't work. while it works from the PSTN phone. similar issue: http://www.voiceie.com/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=8;t=000944 -- Smile, you'll save someone else's day! Frog -- Smile, you'll save someone else's day! Frog
[OSL | CCIE_Voice] CallManager Annunciator
Hi all, When a CallManager's ip phone dialed a unallocated DN#, the annunciator will play the following prompt to the calling ip phone: Your call cannot be completed as dialed. Please consult your directory and call again or ask your operator for assistance. This is a recording. I am just wondering, can I make the annunciator to play the similar prompt when the calling party is external PSTN? Thanks.
[OSL | CCIE_Voice] Multicast Music On Hold: Remote site uses MoH from local SRST flash
Hi everyone, Referring to this Cisco document ( http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_feature_guide09186a00802d1c31.html#wp1118060), there stated The last Cisco CallManager configuration task involves creating an MOH region that assigns MOH G.711 codec usage for the central site or sites and branch office or offices. So, where I should assign the MOH G.711 region? Assuming the requirement for HQ ip phone to remote site ip phone call must use G.729. Thanks.
Re: [OSL | CCIE_Voice] Route Patterns and block patterns
Oh, Yes!! I am doing that as best practice in real life. This is for fraud prevention. But, worse practice for ccie lab :-p As we want to save as much time as possible during lab. HTH On Thu, Apr 3, 2008 at 4:59 PM, Paul and Bobs [EMAIL PROTECTED] wrote: Guys Got a silly question thats bugging me. Do you have to explicitly block patterns you do not want users to be able to dial. Let me tell you what im thinking. 'PhoneA' in in PAR-INT with CSS-ALL 'PhoneB' in in PAR-INT with CSS-INT 1800XXX is in PAR-FREE and only CSS-ALL contains PAR-FREE now I know that only 'PhoneA' can call this as 'PhoneB' does not have css to call but I was wandering if its worth while creating a block pattern for 1800XXX and putting it into PAR-BLOCK and in CSS-INT Hope this all makes sense. Thanks Paul
[OSL | CCIE_Voice] Frame-relay adaptive-shaping
Hi all, Just wondering, is no frame-relay adaptive-shaping interface-congestion required for a frame-relay with FRF.12? Or it doesn't matter if my mincir is equal to cir? Thanks.
Re: [OSL | CCIE_Voice] CME Hunt groups
I will customized the timeout value, as default is 180s ! ephone-hunt 1 sequential pilot 3111 list 3100,3101 final timeout 6 ! HTH On Wed, Apr 2, 2008 at 2:03 PM, Paul and Bobs [EMAIL PROTECTED] wrote: Howdy When creating hunt groups and haveing the hunt group hunt to phone 1 on say a second line and then phone 2 on the second line and finally forwarding to voice mail. Would the following config work telephony-service voicemail ! ephone-hunt 1 sequential pilot 3111 list 3100,3101 final
Re: [OSL | CCIE_Voice] Regions and Locations
Hi paul bobs, You sure you can configure AUDIO bandwidth in Region? You only allowed to choose the AUDIO codec in Region and configure VIDEO Call bandwidth in Region. However, you can configure AUDIO bandwidth in Location. So, if you select audio codec g711 in Region and configure audio bandwidth to 80 in Location, then you are allowing only 1 x g711 call. HTH, On Thu, Mar 27, 2008 at 2:21 PM, Paul and Bobs [EMAIL PROTECTED] wrote: Whats the difference between setting the audio bandwidth to 80 in the region and setting the location bandwidth to 80 for bandwidth management over the WAN
[OSL | CCIE_Voice] FXO Disconnect problem
Hi, Wondering anyone faced fxo disconnect problem before? pstn -- telco -- CO Trunk -- FXO -- Router -- VOIP -- CallManager -- ephone 1 So, my issue is when pstn calling to ephone 1 and hang up before the call is answered. But ephone 1 will keep ringing till the ephone 1 goes off hock. I have the following configured for my fxo port: ! ! voice-port 0/1/0 supervisory disconnect dualtone pre-connect timeouts call-disconnect 1 ! ! It is a Cisco 2851 and vic2-4fxo running IOS 12.4(8a) Thanks.
Re: [OSL | CCIE_Voice] CME Version
Should be still 3.3 On Thu, Mar 27, 2008 at 8:56 PM, Christian Narvaez [EMAIL PROTECTED] wrote: Guys, For those who have recently taken the practical exam. Which CallManager Express Version has been used 3.3, 4.0 or 4.1 ?? IOS version 12.4supports all them. PD: Hope I am not trying to violate any NDA :P
Re: [OSL | CCIE_Voice] ipcc express task 10.1 call connected but no sound.
Well, This is quite common issue using ip blue softphone. I had the similar issue while using ip blue for ipcc and unity. So, I now use CIPC for ipcc and unity testing. On Sun, Mar 23, 2008 at 9:37 AM, [EMAIL PROTECTED] wrote: I am using ipblue softphone DN1003 with DP,location HQ same as route point. when made call to route point 1700, call can connected but there is no sound...:( call to other phones 1002 etc is working fine. anyone experience same problem? i am using pod12 how can i troubleshoot this ? Sara -- Easy + Joy + Powerful = Yahoo! Bookmarks x Toolbarhttp://pr.mail.yahoo.co.jp/toolbar/
Re: [OSL | CCIE_Voice] IP blue multiple instance video link
from command prompt, vtgo-pc.exe /d well, you got to navigate to ip blue directory first before issuing that command. On Sun, Mar 23, 2008 at 11:06 AM, [EMAIL PROTECTED] wrote: may i know where is the video link to turn on multiple instance of ipblue? Sara -- Easy + Joy + Powerful = Yahoo! Bookmarks x Toolbarhttp://pr.mail.yahoo.co.jp/toolbar/
Re: [OSL | CCIE_Voice] Whats Missing
Give the lab a try!! And you will know what is missing. Well, i personally think it might be better to complete within 6 - 7 hrs so you can have time for troubleshooting. On Thu, Mar 20, 2008 at 6:32 PM, Edward French [EMAIL PROTECTED] wrote: I have seen several comments about people who have attempted the test in the past couple of weeks. How well did the Proctor labs ultimate lab guide prepare you for the test? What areas did you find the most difficult? I can quickly and without reference perform all tasks in the books with the exception of: IPMA, Fast/Quick Dial, EM, Fax, BACD, QOS on 6500, QOS on FR,and sometimes Gatekeeper gets me. I can quickly find the IPMA, Fast/Quick Dial, EM, Fax and BACD on the univercd or other available source. and I can usually complete the full lab scenarios in th 7:45 proctor lab session. Based on your experience with the lab and my above statements do you think I am ready to take the lab? Additionally I have been working in voice for 21 years, I have been a CCNA for I think 10 years and I have been Microsoft certified since NT. Thanks for your opinions Ed
Re: [OSL | CCIE_Voice] Can't do restore factory default on CUE
So, there is no way I can restore my CUE 2.0.1 to factory default? Thanks. On Wed, Mar 19, 2008 at 11:13 PM, John [EMAIL PROTECTED] wrote: You're using Unity express version 2.0.1. The restore to factory defaults was not added until version 2.1 according to the 2.1 release notes ( http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/rel2_1/relnote/RNcue21.html). You might want to upgrade your CUE module and start from there. John *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *ccievoice1 *Sent:* Wednesday, March 19, 2008 8:10 AM *To:* CCIE Maillist *Subject:* [OSL | CCIE_Voice] Can't do restore factory default on CUE Hi all, I am not able to perform a restore factory defualt on my cue. Any advice? se-142-102-66-253(offline) restore ? id ID number of the backup se-142-102-66-253(offline) restore factory default restore factory default ^ % Invalid input detected at '^' marker. se-142-102-66-253(offline) ! ! se-142-102-66-253 show version CPU Model:Celeron (Coppermine) CPU Speed (MHz): 298.899 CPU Cache (KByte):128 Chassis Type: C3725 Chassis Serial: JHY0843K0HJ Module Type: AIM Module Serial:FOC090443PD ! ! se-142-102-66-253 sh software versions Installed Packages: - Bootloader (Primary) 1.0.17 - Global 2.0.1 - Voice Mail 2.0.1 - Bootloader (Secondary) 2.0.1 - Core 2.0.1 - Installer 2.0.1 - Auto Attendant 2.0.1 Installed Languages: - US English 2.0.1 ! ! Thanks.
Re: [OSL | CCIE_Voice] Lab running on VMWare
I am for CCM, IPCC Express and Unity!! On Mon, Mar 17, 2008 at 10:04 PM, Scott Monasmith [EMAIL PROTECTED] wrote: Is anyone running their lab equipment in VMWare?
Re: [OSL | CCIE_Voice] IPMA - Wizard or Manual Configuration
Well, Ben Ng once said you can use auto-qos actually!! And, IPMA wizard if you're comfortable with it!! On Tue, Mar 18, 2008 at 12:38 AM, Jonathan Charles [EMAIL PROTECTED] wrote: Cisco's trend on the lab is that if there is an automagical way to do it, it is forbidden... So, no auto qos, no voicemail wizard, no ipma wizard, no dial plan wizard, no setup on the routers... Jonathan On Mon, Mar 17, 2008 at 11:28 AM, Christian Narvaez [EMAIL PROTECTED] wrote: Guys, Which is the best method of configuring IPMA in the real lab. Via Wizard or Manually ? Currently I am focusing in learning the Wizard, but I dont know If you have seen any contrain based on your experience. Thanks Christian Narvaez
Re: [OSL | CCIE_Voice] Lab running on VMWare
Lots of FAAAST hardisk space :-) On Tue, Mar 18, 2008 at 1:24 AM, Jonathan Charles [EMAIL PROTECTED] wrote: Bigger is better, you want 4GB of RAM, lots of CPU and lots of HD space... Jonathan On Mon, Mar 17, 2008 at 12:05 PM, Vik Ahuja [EMAIL PROTECTED] wrote: What's the minimum box recommended with regards to CPU, memory, etc for Vmware to run efficiently?. Thanks guys Jonathan Charles [EMAIL PROTECTED] wrote: Did you install VMWare tools? J On Mon, Mar 17, 2008 at 9:29 AM, Balamurugan Singaram wrote: Hi, I am running VMWare in windowsXP, my host have sound card installed, and sound card is shown as connected to vmware, but I am not able to access the soundcard in vmware win 2k3 server. Can please let me know what I am missing. Thanks, Bala. ccievoice1 wrote: I am for CCM, IPCC Express and Unity!! On Mon, Mar 17, 2008 at 10:04 PM, Scott Monasmith wrote: Is anyone running their lab equipment in VMWare? Send instant messages to your online friends http://uk.messenger.yahoo.com
[OSL | CCIE_Voice] OT: Error accessing to www.proctorlabs.com
Hi, Anyone having problem to access proctorlabs website? I can't access at all.. Is proctorlabs hit by the recent massive web attacks which attacked McAfee and Trend Micro as well? Would be good if Proctorlabs/ IPExpert support able to lets us know when the service will be resume. Thanks.
Re: [OSL | CCIE_Voice] QOS 6500
Just one comment, H323 RAS is using UDP for port 1718 and 1719 Also, since the task is asking to remark the voice signaling traffic from CallManager on Catalyst6500, hence I would define all the qos acl on the Catalyst6500 to do the job. Thanks. On Mon, Mar 17, 2008 at 7:00 AM, Edward French [EMAIL PROTECTED] wrote: Lab 25 task 46 says to Configure the Catalyst 6500 to mark all VOIP control traffic from the CallManager as AF31. Which would be: set port qos 3/23 port-based set qos acl ip SERVER dscp 26 tcp any range 2000 2002 any set qos acl ip SERVER dscp 26 tcp any any range 11000 11999 set qos acl ip SERVER dscp 26 tcp any any range 1024 4999 set qos acl ip SERVER dscp 26 tcp any any range 1719 1720 set qos acl ip SERVER dscp 26 udp any eq 2427 any set qos acl ip SERVER dscp 26 tcp any eq 2428 any commit qos acl SERVER set qos acl map SERVER 3/23 And in the DVD and Audio it says the best thing to do is copy the QOS commands from the SRND. However the above commands are not in the SRND. The SRND say do the following set port qos 3/23 trust trust-dscp or on a 2Q2T port set qos acl ip TRUST-DSCP trust-dscp any commit qos acl TRUST-DSCP set qos acl map TRUST-DSCP 3/23 My question is do the commands from the SRND meet the requirements of the task? And if not is there an easier way than the first solution, since this ACL is not in the SRND? Also given that the first solution is an older solution is this what I should expect on the lab? Basically what is the easiest way to find and copy the QOS statements for the 6500? I do not have access to one outside of the lab so I would appreciate any tricks or tips rather than memorization. Thanks Ed
[OSL | CCIE_Voice] Catalyst 6500 QoS Marking
! ! set qos cos-dscp-map 0 8 16 24 32 46 48 56 set qos policed-dscp-map 0,24,26,46:8 set qos policer aggregate VVLAN-VOICE rate 128 burst 8000 drop set qos policer aggregate VVLAN-CALL-SIGNALING rate 32 burst 8000 policed-dscp set qos acl ip IPPHONE dscp 46 aggregate VVLAN-VOICE udp 177.1.101.0 255.255.255.0 any range 16384 32767 set qos acl ip IPPHONE dscp 24 aggregate VVLAN-CALL-SIGNALING tcp 177.1.101.0 255.255.255.0 any range 2000 2002 commit qos acl IPPHONE set port qos 2/45 trust-device ciscoipphone set qos acl map IPPHONE 2/45 ! ! Question: on policer aggregate, burst is to specifies the burst size; valid values are 1 to 32000 kilobits. So what value do we use? And how to calculate the value? Thanks.
[OSL | CCIE_Voice] 3550 QoS : Policing and Marking
Hi all, I have configured 3550 for policing and remarking on Cisco IP Phone ports. Question, do I need to configure the uplink-to-router port to trust the dscp value? Thanks.
Re: [OSL | CCIE_Voice] 3550 QoS : Policing and Marking
Are cisco 3550 that fickle? Perhaps not :-) Thanks. On Sat, Mar 15, 2008 at 12:24 PM, Jonathan Charles [EMAIL PROTECTED] wrote: Only access ports clear DSCP markings... so, you shouldn't need to re-trust (besides, you just trusted them less than a port ago... are your switches that fickle?) Then again, are they that fickle? My guess would be 'no'... but I have been wrong 46 times just today... Jonathan On Fri, Mar 14, 2008 at 11:16 PM, ccievoice1 [EMAIL PROTECTED] wrote: Hi all, I have configured 3550 for policing and remarking on Cisco IP Phone ports. Question, do I need to configure the uplink-to-router port to trust the dscp value? Thanks.
[OSL | CCIE_Voice] CME: DND
Hi all, Just wondering is it possible when 2111 calling , and pressed DND then 2111 will hear only silence? ! ephone-dn 2 number ! ephone 2 button 1f2 no dnd feture-ring -- and exactly what this command do? ! Thanks.
Re: [OSL | CCIE_Voice] VM in SRST mode
Hi Vik, Can I assume during *normal* srst failover, vm-integration is not required. However, due to IOS bug for a certain IOS version, vm-integration is required. Am I right to say that? Thanks. On Wed, Feb 20, 2008 at 1:19 PM, anil batra [EMAIL PROTECTED] wrote: yes that's correct..1599 is Hunt Pilot --- Vik Malhi [EMAIL PROTECTED] wrote: I assume- 1599 is your hunt pilot to voicemail. The vm-integration command only takes effect when the call-forward # = the voicemail #. So add this command: Call-manager-fallback voicemail 12122241599 Vik Vik Malhi - CCIE #13890, CCSI #31584 Sr Technical Instructor - IPexpert, Inc. A Cisco Learning Partner - We Accept Learning Credits! Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of anil batra Sent: Tuesday, February 19, 2008 8:41 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] VM in SRST mode Hi Vik I am folllowing the configs for VM in SRST. I used VM integration but still it goes to Unity pening greeting instead of the greeting for 2001. Under noraml it hits VM of 2001. Here is the debug and config on Br1 - P24-BR1-RTR# .Feb 20 04:50:30.397: ISDN Se0/0/0:23 Q931: RX - SETUP pd = 8 callref = 0x0097 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, '2122241004' Plan:ISDN, Type:National Called Party Number i = 0xA1, '6175242001' Plan:ISDN, Type:National .Feb 20 04:50:30.413: ISDN Se0/0/0:23 Q931: TX - CALL_PROC pd = 8 callref = 0x8097 Channel ID i = 0xA98381 Exclusive, Channel 1 P24-BR1-RTR# .Feb 20 04:50:30.425: ISDN Se0/0/0:23 Q931: TX - ALERTING pd = 8 callref = 0x8097 Progress Ind i = 0x8188 - In-band info or appropriate now available P24-BR1-RTR# .Feb 20 04:50:36.421: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type 0xD is 0x2 0x1, Calling num 2122241004 .Feb 20 04:50:36.425: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type 0xD is 0x2 0x1, Called num 12122241599 .Feb 20 04:50:36.425: ISDN Se0/0/0:23 Q931: TX - SETUP pd = 8 callref = 0x0084 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98383 Exclusive, Channel 3 Calling Party Number i = 0x2181, '2122241004' Plan:ISDN, Type:National Called Party Number i = 0xA1, '12122241599' Plan:ISDN, Type:National Redirecting Number i = 0xFF, '2001' Plan:Reserved, Type:Reserved .Feb 20 04:50:36.457: ISDN Se0/0/0:23 Q931: RX - CALL_PROC pd = 8 callref = 0x8084 Channel ID i = 0xA98383 Exclusive, Channel 3 .Feb 20 04:50:36.493: ISDN Se0/0/0:23 Q931: RX - ALERTING pd = 8 callref = 0x8084 Progress Ind i = 0x8088 - In-band inf P24-BR1-RTR#o or appropriate now available .Feb 20 04:50:36.693: ISDN Se0/0/0:23 Q931: RX - CONNECT pd = 8 callref = 0x8084 Display i = 'Voicemail' .Feb 20 04:50:36.697: %ISDN-6-CONNECT: Interface Serial0/0/0:2 is now connected to 12122241599 N/A .Feb 20 04:50:36.697: ISDN Se0/0/0:23 Q931: TX - CONNECT_ACK pd = 8 callref = 0x0084 .Feb 20 04:50:36.705: ISDN Se0/0/0:23 Q931: TX - CONNECT pd = 8 callref = 0x8097 .Feb 20 04:50:36.713: ISDN Se0/0/0:23 Q931: RX - CONNECT_ACK pd = 8 callref = 0x0097 .Feb 20 04:50:36.717: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now connected to 2122241004 N/A P24-BR1-RTR# .Feb 20 04:50:40.940: ISDN Se0/0/0:23 Q931: RX - DISCONNECT pd = 8 callref = 0x0097 Cause i = 0x8290 - Normal call clearing .Feb 20 04:50:40.944: %ISDN-6-DISCONNECT: Interface Serial0/0/0:0 disconnected from 2122241004 , call lasted 4 seconds .Feb 20 04:50:40.944: ISDN Se0/0/0:23 Q931: TX - RELEASE pd = 8 callref = 0x8097 .Feb 20 04:50:40.952: %ISDN-6-DISCONNECT: Interface Serial0/0/0:2 disconnected from 12122241599 , call lasted 4 seconds P24-BR1-RTR# .Feb 20 04:50:40.952: ISDN Se0/0/0:23 Q931: TX - DISCONNECT pd = 8 callref = 0x0084 Cause i = 0x8290 - Normal call clearing .Feb 20
[OSL | CCIE_Voice] Route Group Failover
Hi all, I have created 2 Route List: *RL_HQ_Local* and *RL_SiteB_Local *Within the RL_HQ_Local, I have: RG_6608 -- 6608 T1 PRI Gateway (1st priority) RG_H323 -- H323 T1 PRI Gateway (2nd priority) Within the RL_SiteB_Local, I have RG_H323 -- H323 T1 PRI Gateway (1st priority) RG_6608 -- 6608 T1 PRI Gateway (2nd priority) For Scenario 1. RL_HQ_Local the call is able to failvoer to RG_H323 if RG_6608 is not reachable. Scenario 2. RL_SiteB_Local call does not able to failover to RG_6608 when RG_H323 failed. So, is there any paramaters required in CallManager for scenario 2 failover? Thanks.
Re: [OSL | CCIE_Voice] Route Group Failover
Hmm, Just to reconfirm, Route pattern of 521 --Route List -- H323 route group --siteb h323 gateway -- 6608 route group -- 6608 pri gateway So, if call to 521 *failed on* first route group, it should able to hunt to next route group available in the route list, right? Thanks. On Feb 16, 2008 10:15 PM, Ovais Iqbal [EMAIL PROTECTED] wrote: Yes please explain how r u simulating an outage to test the 6608 as a backup and also provide details on the digit manipulation are u doing and where. Ovais Iqbal 416-294-7869 Sent from my BlackBerry device -Original Message- From: Jose Linero Welcker [EMAIL PROTECTED] Date: Sat, 16 Feb 2008 13:51:58 To:boonchin .ng [EMAIL PROTECTED] Cc:CCIE Maillist ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Route Group Failover Hi How are you doing the test?, could you post the h323 debug. Regards, Jose Date: Sat, 16 Feb 2008 21:42:36 +0800 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Route Group Failover In the RG. I did test calling the particular destination number to specific route group at a time and the call succeed. So the route-pattern, route list, route group, gateway and h323' dial-peer are configured correctly. Only when I assign H323 and 6608 in a single route list, the call is not able to failover from h323 to 6608 Thanks. On Feb 16, 2008 9:25 PM, Jose Linero Welcker [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi: No there is not an specific requirement, where are you doing the digit manipulation, in the RP or in the RG? Regards, Jose Date: Sat, 16 Feb 2008 19:38:12 +0800 From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] To: ccie_voice@onlinestudylist.com mailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Route Group Failover Hi all, I have created 2 Route List: RL_HQ_Local and RL_SiteB_Local Within the RL_HQ_Local, I have: RG_6608 -- 6608 T1 PRI Gateway (1st priority) RG_H323 -- H323 T1 PRI Gateway (2nd priority) Within the RL_SiteB_Local, I have RG_H323 -- H323 T1 PRI Gateway (1st priority) RG_6608 -- 6608 T1 PRI Gateway (2nd priority) For Scenario 1. RL_HQ_Local the call is able to failvoer to RG_H323 if RG_6608 is not reachable. Scenario 2. RL_SiteB_Local call does not able to failover to RG_6608 when RG_H323 failed. So, is there any paramaters required in CallManager for scenario 2 failover? Thanks. Express yourself instantly with MSN Messenger! MSN Messenger http://clk.atdmt.com/AVE/go/onm00200471ave/direct/01/ Express yourself instantly with MSN Messenger! MSN Messenger
Re: [OSL | CCIE_Voice] Route Group Failover
Yeah, I am looking the the Stop Routing on Unallocated Number Flag* Thanks everyone for the input. On Feb 17, 2008 12:53 AM, Burkett, Michael [EMAIL PROTECTED] wrote: If you are shutting the interface down to test the gateway will send an unallocated/unassigned message back the CCM, the default behavior on that message is to stop routing. Look in CCM service parameters and change the setting. Stop Routing on Unallocated Number Flag* is set to True by default, set to false and you will move to the next member in the Route List. *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Jose Linero Welcker *Sent:* Saturday, February 16, 2008 10:48 AM *To:* ccievoice1; [EMAIL PROTECTED] *Cc:* CCIE Maillist; [EMAIL PROTECTED] *Subject:* Re: [OSL | CCIE_Voice] Route Group Failover Hi: Yes that is the way it should work, however depending on how you are simulating an outage, the CCM knows there is a failed call and try to reroute the call on the second route group. Depending on the H323 message the callmanager is receiving it can reroute the call, please tell how are you simulating the outage, turn the E1 down?, turn the gateway down? Regards, Jose -- Date: Sat, 16 Feb 2008 22:23:52 +0800 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] CC: ccie_voice@onlinestudylist.com; [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] Route Group Failover Hmm, Just to reconfirm, Route pattern of 521 --Route List -- H323 route group --siteb h323 gateway -- 6608 route group -- 6608 pri gateway So, if call to 521 *failed on* first route group, it should able to hunt to next route group available in the route list, right? Thanks. On Feb 16, 2008 10:15 PM, Ovais Iqbal [EMAIL PROTECTED] wrote: Yes please explain how r u simulating an outage to test the 6608 as a backup and also provide details on the digit manipulation are u doing and where. Ovais Iqbal 416-294-7869 Sent from my BlackBerry device -Original Message- From: Jose Linero Welcker [EMAIL PROTECTED] Date: Sat, 16 Feb 2008 13:51:58 To:boonchin .ng [EMAIL PROTECTED] Cc:CCIE Maillist ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Route Group Failover Hi How are you doing the test?, could you post the h323 debug. Regards, Jose Date: Sat, 16 Feb 2008 21:42:36 +0800 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Route Group Failover In the RG. I did test calling the particular destination number to specific route group at a time and the call succeed. So the route-pattern, route list, route group, gateway and h323' dial-peer are configured correctly. Only when I assign H323 and 6608 in a single route list, the call is not able to failover from h323 to 6608 Thanks. On Feb 16, 2008 9:25 PM, Jose Linero Welcker [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi: No there is not an specific requirement, where are you doing the digit manipulation, in the RP or in the RG? Regards, Jose Date: Sat, 16 Feb 2008 19:38:12 +0800 From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] To: ccie_voice@onlinestudylist.com mailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Route Group Failover Hi all, I have created 2 Route List: RL_HQ_Local and RL_SiteB_Local Within the RL_HQ_Local, I have: RG_6608 -- 6608 T1 PRI Gateway (1st priority) RG_H323 -- H323 T1 PRI Gateway (2nd priority) Within the RL_SiteB_Local, I have RG_H323 -- H323 T1 PRI Gateway (1st priority) RG_6608 -- 6608 T1 PRI Gateway (2nd priority) For Scenario 1. RL_HQ_Local the call is able to failvoer to RG_H323 if RG_6608 is not reachable. Scenario 2. RL_SiteB_Local call does not able to failover to RG_6608 when RG_H323 failed. So, is there any paramaters required in CallManager for scenario 2 failover? Thanks. Express yourself instantly with MSN Messenger! MSN Messenger http://clk.atdmt.com/AVE/go/onm00200471ave/direct/01/ Express yourself instantly with MSN Messenger! MSN Messenger -- Express yourself instantly with MSN Messenger! MSN Messengerhttp://clk.atdmt.com/AVE/go/onm00200471ave/direct/01/ ~ This email message is for the sole use of the intended recipient(s) and may contain confidential and privileged information of Cameron and its Operating Divisions. Any unauthorized use or disclosure is prohibited. If you are not the intended recipient, please contact the sender by reply email and delete and destroy all copies of the original message inclusive of any attachments. ~
[OSL | CCIE_Voice] OT: Proctorlabs Voice vRack' PSTN Phone
Hi, I followed https://proctorlabs.com/downloads/media/PSTN-IPBlue-Demo.mp4, and noticed there is a list of fix-mac-address that I am required to assign to my pstn softphonephone depending to the pod i am assigned to. Just wondering, where can I get the fix-mac-addresses list? Thanks.
[OSL | CCIE_Voice] QoS question
Hi, I have a 384K frame-relay link. Hence, I have: ! map-class frame-relay FRTS-384 frame-relay cir 364800 frame-relay bc 3648 frame-ralay be 0 frame-relay mincir 364800 frame-relay fragment 480 ! Now, I want to do LLQ on voice, so: ! policy-map LLQ class media priority percent 33 class control bandwidth percent 5 class class-default fair-queue ! *Question*: In the policy-map LLQ, I have bandwidth percent 5. So can I assume that is 5% of the CIR value? If I want to use priority x and bandwidth y without using the *percent command*, how I can calculate the bandwidth value for class media and class control? Thanks.
[OSL | CCIE_Voice] Intermittent one-way voice issue
Hi all, Just wondering what might caused *intermittent one-way voice issue *for an incoming pstn call to CallManager controlled IP Phones? Thanks.
[OSL | CCIE_Voice] OT: Lab Swap - 5th May 2008 Brussels -- April/ May Tokyo
Hi, I am having 5th May Brussels lab date. Am thinking to take my exam in Tokyo. So just wondering if anyone interested to swap, please kindly unicast me Thanks.
[OSL | CCIE_Voice] SIP Analog Gateway and FXS port
1. On IOS Router: voice service voip sip bind control source-interface fas0/0.110 bind media source-interface fas0/0.110 voice-port 2/0/2 station name Analog1 station number dial-peer voice 1 pots application session destination-pattern port 2/0/2 dial-peer voice 10 voip destination-pattern [23]... session protocol sipv2 session target ipv4:10.1.200.21 dtmf-relay rtp-nte codec g711ulaw no vad dial-peer voice 11 voip destination-pattern [23]... session protocol sipv2 session target ipv4:10.1.200.20 dtmf-relay rtp-nte codec g711ulaw no vad preference 1 2. On CallManager Create a SIP trunk poiting to IOS Router' FastEthernet0/0.110 ip address Media Termination Point Required = Checked Assigned the SIP Trunk to device pool with region codec set to G711 So, are the above configuration correct? Also, when should we use *dtmf-relay sip-notify and dtmf-relay rtp-nte*? Does that make any different to CallManager SIP Trunk configuration when using different dtmf-relay? Thanks.
Re: [OSL | CCIE_Voice] SIP Analog Gateway and FXS port
And, of course the appropriate route-pattern in CallManager pointing to the SIP Trunk. Thanks and best regards. On Feb 4, 2008 7:01 PM, ccievoice1 [EMAIL PROTECTED] wrote: 1. On IOS Router: voice service voip sip bind control source-interface fas0/0.110 bind media source-interface fas0/0.110 voice-port 2/0/2 station name Analog1 station number dial-peer voice 1 pots application session destination-pattern port 2/0/2 dial-peer voice 10 voip destination-pattern [23]... session protocol sipv2 session target ipv4:10.1.200.21 dtmf-relay rtp-nte codec g711ulaw no vad dial-peer voice 11 voip destination-pattern [23]... session protocol sipv2 session target ipv4:10.1.200.20 dtmf-relay rtp-nte codec g711ulaw no vad preference 1 2. On CallManager Create a SIP trunk poiting to IOS Router' FastEthernet0/0.110 ip address Media Termination Point Required = Checked Assigned the SIP Trunk to device pool with region codec set to G711 So, are the above configuration correct? Also, when should we use *dtmf-relay sip-notify and dtmf-relay rtp-nte*? Does that make any different to CallManager SIP Trunk configuration when using different dtmf-relay? Thanks.
Re: [OSL | CCIE_Voice] IPCC Question
I think password should be *telecaster* On Jan 30, 2008 11:24 AM, Chad Stachowicz [EMAIL PROTECTED] wrote: I think this one must be simple, but i couldn't find it on cisco documentaiton. Is there a special password for telecaster, or should it be cisco? Chad
