Re: [OSL | CCIE_Voice] Taking the lab next week (Mike Thompson)
Take the rest day before exam and sleep well on the last night, You need to be FRESH on exam day is this your first attempt ? GOOD LUCK From: ccie_voice-requ...@onlinestudylist.com ccie_voice-requ...@onlinestudylist.com To: ccie_voice@onlinestudylist.com Sent: Sat, 25 June, 2011 8:03:06 Subject: CCIE_Voice Digest, Vol 64, Issue 202 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: GradedLabs (INE) rack rental (Bill Lake) 2. Re: GradedLabs (INE) rack rental (Dave) 3. Re: GradedLabs (INE) rack rental (Mike Thompson) 4. Re: Taking the lab next week (Mike Thompson) 5. RES: GradedLabs (INE) rack rental (Marcelo Alexandria) -- Message: 1 Date: Sat, 25 Jun 2011 06:46:22 -0500 From: Bill Lake whl...@gmail.com To: Dave ccvoic...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] GradedLabs (INE) rack rental Message-ID: BANLkTimUc7=cjsfhwyu22nnwuh_epfo...@mail.gmail.com Content-Type: text/plain; charset=ISO-8859-1 Hello, I have never used them but their sessions are 5.5 hours long, but they do seem cheaper and this weekend they have a buy one get one free promo this is the email I got Rack Rentals, Buy One Get One Free! This weekend only! Save over 50% on your next rack rental purchase from INE.com. This offer applies to the following rack rental token bundles: * 500 Rack Rental Tokens - Only $249 a $251 savings! - Buy Now * 700 Rack Rental Tokens - Only $349 a $351 savings! - Buy Now * 1000 Rack Rental Tokens - Only $499 a $501 savings! - Buy Now Rack Rental Token Details Ready to get the hands on experience needed to pass the actual CCIE Lab Exam? Look no further than our CCIE Rack Rentals, powered by Gradedlabs.com. We offer rack rentals with simplified scheduling and advanced management control of CCNA, CCNP, CCIE Routing Switching, CCIE Voice, CCIE Security, and CCIE Service Provider Rack Rentals that support all of our Self-Paced and Instructor-Led product lines. Rack rentals are sold in 5.5 hour blocks and are offered in four time slots: S1. 3:00am - 8:30am Pacific Daylight Time (-7 UTC) S2. 9:00am - 2:30pm Pacific Daylight Time (-7 UTC) S3. 3:00pm - 8:30pm Pacific Daylight Time (-7 UTC) S4. 9:00pm - 2:30am Pacific Daylight Time (-7 UTC) On Fri, Jun 24, 2011 at 11:34 PM, Dave ccvoic...@gmail.com wrote: Hi, Can you comment on gradedlabs (INE) voice rack quality if you have used before? its price is cheaper compared to IPExpert proctorlabs (2.5 times cheaper) CCBootcamp (NLI) seems the most expensive Voice Rack. Can anybody comment on quality if you have used before? Thanks, Dave ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- Message: 2 Date: Sat, 25 Jun 2011 19:01:03 +0530 From: Dave ccvoic...@gmail.com To: Bill Lake whl...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] GradedLabs (INE) rack rental Message-ID: BANLkTimbcvdPxuFTtjBzaVYOM=jtv8g...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi, This bundle has lowest commitment of 33 sessions (of 5.5 hours duration). Before going , I needed some feedback from INE rack users. Dave On Sat, Jun 25, 2011 at 5:16 PM, Bill Lake whl...@gmail.com wrote: Hello, I have never used them but their sessions are 5.5 hours long, but they do seem cheaper and this weekend they have a buy one get one free promo this is the email I got Rack Rentals, Buy One Get One Free! This weekend only! Save over 50% on your next rack rental purchase from INE.com. This offer applies to the following rack rental token bundles: * 500 Rack Rental Tokens - Only $249 a $251 savings! - Buy Now * 700 Rack Rental Tokens - Only $349 a $351 savings! - Buy Now * 1000 Rack Rental Tokens - Only $499 a $501 savings! - Buy Now Rack Rental Token Details Ready to get the hands on experience needed to pass the actual CCIE Lab Exam? Look no further than our CCIE Rack Rentals, powered by Gradedlabs.com. We offer rack rentals with simplified scheduling and advanced management control of CCNA, CCNP, CCIE Routing Switching, CCIE Voice, CCIE Security, and CCIE Service Provider Rack Rentals that support all of our Self-Paced and Instructor-Led product lines. Rack
[OSL | CCIE_Voice] how to find b-acd on cisco documentation
Hi All Can you guid me to find b-acd on cisco documentation. Is any one has the steps link. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] how to find b-acd on cisco documentation
Actually, this is ended with PDF file, can we open PDF file during exam time. From: ccie_voice-requ...@onlinestudylist.com ccie_voice-requ...@onlinestudylist.com To: ccie_voice@onlinestudylist.com Sent: Wed, 22 June, 2011 11:37:28 Subject: CCIE_Voice Digest, Vol 64, Issue 177 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: route group distribution algorithm (Randall Saborio) 2. UCM LOCATION QUESTION (Stephen Manuel) 3. Re: route group distribution algorithm (givemeccievoice2...@gmail.com) 4. Re: UCM LOCATION QUESTION (Cristobal Priego) -- Message: 1 Date: Wed, 22 Jun 2011 10:06:48 -0600 From: Randall Saborio ill2...@gmail.com To: Cristobal Priego cristobalpri...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] route group distribution algorithm Message-ID: BANLkTin5zSzWXV08XPM=+x5je2tspgf...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Or maybe because it is never used or it never matters. If it would matter, then they would be specific about it. On Wed, Jun 22, 2011 at 9:59 AM, Cristobal Priego cristobalpri...@gmail.com wrote: the reason why i am asking is because in the lab they're not that specific 2011/6/22 CCIE STUDENT cciefo...@hotmail.com You rarely even do it in the real world -Original Message- From: Randall Saborio ill2...@gmail.com Date: Wed, 22 Jun 2011 13:26:24 To: cristobalpri...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] route group distribution algorithm Can you name when on the lab would you ever configure two devices on same route group? I don't think this is ever done on the lab. Even if it did, its not a matter of preference, but a matter of matching the task requirements. On Tue, Jun 21, 2011 at 7:41 PM, Cristobal Priego cristobalpri...@gmail.com mailto:cristobalpri...@gmail.com wrote: guys, for the lab, whenever you configure your route groups which distribution algorithm is better circular or top down i use top down all the time, but i'd like to know your opinion on this thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com http://www.PlatinumPlacement.com -- Randall da ill Saborio CCIE Voice Wannabe #10054675811 -- Randall da ill Saborio CCIE Voice Wannabe #10054675811 -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20110622/650f2f6f/attachment-0001.html -- Message: 2 Date: Wed, 22 Jun 2011 13:18:18 -0400 From: Stephen Manuel srman...@bellsouth.net To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] UCM LOCATION QUESTION Message-ID: 007201cc3100$615f7a70$241e6f50$@bellsouth.net Content-Type: text/plain; charset=us-ascii Experts, I have a somewhat basic question just clear up something that is lingering in my mind. The example is this: In a multiple remote location site company, with a Centralized UCM architecture, each site has it's own device pool, region, and location settings. All the DN's on the phones for all sites are in the same PT, meaning calls between sites are on net via their existing WAN infrastructure and not the SIP Trunk/PSTN. Remote site A is rather large 50-60 phones. Remote site B is rather small 10-15 phones. Calls between regions use g729 Calls to the PSTN using a SIP Trunk use g711 The location setting for site A is unlimited. The location setting for site B is 256 What I am wondering is this, Site B has 3 calls to the PSTN using the SIP Trunk using g711 @ 80k per call which equals 240k. The SIP trunk is also in it's own Device Pool, region, etc. Now someone from Site A calls site B, the bandwidth at Site A is unlimited so no issue there. According to the Cisco Research I've done, a call requires 2 streams. Since Site B is using 240 of 256k of bandwidth on the 3 PSTN/SIP Trunk calls, when a g729 calls comes inbound which requires 24k which causes Site B location setting to be exceeded, will the call fail or succeed ?? IMO, the call will fail. Thanks, Stephen Manuel -- next part -- An HTML attachment was
Re: [OSL | CCIE_Voice] RSVP bandwidth allocation on LAN?
How did you configure your agent. Can you submit your configuration? From: ccie_voice-requ...@onlinestudylist.com ccie_voice-requ...@onlinestudylist.com To: ccie_voice@onlinestudylist.com Sent: Sat, 18 June, 2011 20:40:52 Subject: CCIE_Voice Digest, Vol 64, Issue 157 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. RSVP bandwidth allocation on LAN? (Greg) 2. Re: NTP summer time (Bill Lake) 3. Total bandwidth from multiple PVCs on serialinterface (Vega Wong) 4. Re: sip phone register (Bill Lake) 5. Re: NTP summer time (Rrcrumm) -- Message: 1 Date: Sat, 18 Jun 2011 20:13:44 -0500 From: Greg cnn...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] RSVP bandwidth allocation on LAN? Message-ID: BANLkTikH33Rf7DkG4L8uwb8_DHi=jby...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 I configured RSVP agent on routers for Call Admission Control with CUCM. Here's the screen output when there's a call connected: R1#show ip rsvp interface interfacersvp allocated i/f max flow max sub max Gi0/0ena 0 750M 750M 0 Gi0/0.102ena 0 112K 112K 0 Se0/1/0 ena 24K1158K1158K0 Se0/1/0.103 ena 24K1158K1158K0 For testing purpose, I would like to see bandwidth allocation on the LAN interface. How shall I configure the router to achieve this? Or it's impossible? Thanks! -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20110618/f8a85545/attachment-0001.html -- Message: 2 Date: Sat, 18 Jun 2011 21:12:01 -0500 From: Bill Lake whl...@gmail.com To: Adil Shaikh adil.sha...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] NTP summer time Message-ID: banlktinpmlnpmv9_iqfujltg9i3+i6q...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Clock timezone PST -8 would set the time to -8 hours from GMT and the PST stands for Pacific Standard Time. clock summer-time PDT recurring would add the hour used for Daylight savings time, in this case Pacific Daylight Time The only issue on this is knowing that they changed the dates this is active so you could be in trouble if your graded and the IOS does not do this properly automatically like it should. So recently in the US the start and end of Daylight Savings Time has changed and I do not believe that is reflected in the IOS used in the lab. Therefore you might want to know how to change dates to the proper ones before taking the exam. So for EST in US clock summer-time EDT recurring 2 SUN Mar 02:00 first SUN Nov 02:00 On Sat, Jun 18, 2011 at 5:56 PM, Adil Shaikh adil.sha...@gmail.com wrote: Hi All, Question regarding NTP summer time. I configure summertime as per IPexpert material: clock timezone EST -8 clock summer EST recurr However, many places i have seen this to be configured as: clock timezone EST -8 clock summer EDT recurr Which one shoud be done in exam? Any clue? thanks -adil -- .. . . _7___|___|_|_|adil.sha...@gmail.com . . ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20110618/425f5492/attachment-0001.html -- Message: 3 Date: Sun, 19 Jun 2011 12:45:33 +1000 From: Vega Wong v...@iinet.net.au To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Total bandwidth from multiple PVCs on serialinterface Message-ID: 000301cc2e2a$f5d7da90$e1878fb0$@net.au Content-Type: text/plain; charset=us-ascii Hi I am trying to setup up two frame relay PVCs on a serial interface (WIC-1T). On one PVC, I will have a bandwidth of 1536kbps, and 768kbps on the other one. This gets me thinking that the total bandwidth would be 2304kbps (2.25Mbps). I may be wrong, but I thought on a serial interface - WIC-1T, the bandwidth is 1.544Mbps? Then how does it works with the combined bandwidth from the two PVCs? Please help Cheers Vega -- next part -- An HTML attachment was scrubbed... URL:
Re: [OSL | CCIE_Voice] CCIE_Voice Digest,HQ Phones not Registering
If the IP Phones are getting IP address, then Disable the security on IP Phones then register them Manually. From: ccie_voice-requ...@onlinestudylist.com ccie_voice-requ...@onlinestudylist.com To: ccie_voice@onlinestudylist.com Sent: Thu, 16 June, 2011 8:55:42 Subject: CCIE_Voice Digest, Vol 64, Issue 136 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. Q.931 cause code vs. Q.850 cause code (Greg) 2. HQ Phones not Registering (Vinay Kumar6) 3. Re: Q.931 cause code vs. Q.850 cause code (Chris Martin) 4. Re: HQ Phones not Registering (Randall Crumm) 5. Re: HQ Phones not Registering (shabeeb mohammed) 6. Re: HQ Phones not Registering (Deepak sidana) -- Message: 1 Date: Thu, 16 Jun 2011 09:07:41 -0500 From: Greg cnn...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Q.931 cause code vs. Q.850 cause code Message-ID: banlktikfomqkwfubr5q8ptgolvewg0o...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 In troubleshooting, we'll use cause code to determine the possible issue. But I'm confused when to use Q.931 cause code table versus Q.850 cause code table. For example, debug cch323 h225 generated the following output: May 24 07:49:37.854 PDT: //3/00421B080200/H323/cch323_h225_send_release: Cause = 65; Location = 0 For the Cause = 65, which table shall we look at? Thanks! -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20110616/ba59d52a/attachment-0001.html -- Message: 2 Date: Thu, 16 Jun 2011 20:09:26 +0530 From: Vinay Kumar6 vinayjaisw...@in.ibm.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] HQ Phones not Registering Message-ID: ofdcc6bfae.5ab2242d-on652578b1.00505434-652578b1.00508...@in.ibm.com Content-Type: text/plain; charset=us-ascii Hi, Facing difficulty in registering the HQ Phones to CUCM, Routing is enabled on the HQ router and Vlans are created on the switch. Phones are able to get the IP address from the CUCM (DHCP) server., Not sure what exactly to check for. Warm Regards, Vinay Kumar -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20110616/971b92be/attachment-0001.html -- Message: 3 Date: Thu, 16 Jun 2011 09:48:30 -0500 From: Chris Martin clm.c...@gmail.com To: Greg cnn...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Q.931 cause code vs. Q.850 cause code Message-ID: banlktimfug1wzfxw-48kajyvwjz1jz3...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Codec mismatch, can review the cause codes here: http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/voice_troubleshooting/old/vts_appa.html Chris On Thu, Jun 16, 2011 at 9:07 AM, Greg cnn...@gmail.com wrote: In troubleshooting, we'll use cause code to determine the possible issue. But I'm confused when to use Q.931 cause code table versus Q.850 cause code table. For example, debug cch323 h225 generated the following output: May 24 07:49:37.854 PDT: //3/00421B080200/H323/cch323_h225_send_release: Cause = 65; Location = 0 For the Cause = 65, which table shall we look at? Thanks! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20110616/f41f8d72/attachment-0001.html -- Message: 4 Date: Thu, 16 Jun 2011 08:34:35 -0700 From: Randall Crumm randall.cr...@flextronics.com To: Vinay Kumar6 vinayjaisw...@in.ibm.com, ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] HQ Phones not Registering Message-ID: 4e4c71316199b746aa3da652cf56cc5f0beee...@amsacex3.americas.ad.flextronics.com Content-Type: text/plain; charset=us-ascii Can you manually add them and see an IP address on CUCM list of devices? Randall From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vinay Kumar6 Sent: Thursday, June 16, 2011 7:39 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] HQ Phones not Registering Hi, Facing difficulty in registering the HQ Phones to
Re: [OSL | CCIE_Voice] Looking for serious study partner
I am looking too, lets have a chat then From: ccie_voice-requ...@onlinestudylist.com ccie_voice-requ...@onlinestudylist.com To: ccie_voice@onlinestudylist.com Sent: Thu, 16 June, 2011 6:33:38 Subject: CCIE_Voice Digest, Vol 64, Issue 135 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. Looking for serious study partner (Paul Dardinski) 2. CUPC - hit and miss (Adil Shaikh) 3. Re: CUPC - hit and miss (Adil Shaikh) 4. Re: MWI is not working in CME (ccieid1ot) -- Message: 1 Date: Wed, 15 Jun 2011 20:51:44 -0400 From: Paul Dardinski pa...@marshallcomm.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Looking for serious study partner Message-ID: faa9add5b9aaa344b429eced83560fa0e01...@mcc-s-ops-exch1.ma-inc.com Content-Type: text/plain; charset=us-ascii If anyone is scheduled for voice lab attempt, let me know if interested in studying. Thanks, Paul (#16842 RS/Sec) === Paul Dardinski - CCIE #16842 (RS Security) CCVP, CCNP, CCDA, MCSE, MBA Cisco Wireless Specialist Marshall Communications 20098 Ashbrook Place Suite 260 Ashburn, VA 20147 (571) 209-3905 FAX: (571) 223-2012 === From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Greg Sent: Wednesday, June 15, 2011 4:18 PM To: Adam Frankel (afrankel) Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] how to verify class-based traffic shape? I cannot have it on interface since I have MQC class configured. The system gave me an error if I tried to. On Wed, Jun 15, 2011 at 3:11 PM, Adam Frankel (afrankel) afran...@cisco.com wrote: Do you have frame-relay trafic-shaping on the physical serial interface? (not the sub-interface) Adam Original Message-- From: Greg cnn...@gmail.com mailto:cnn...@gmail.com Sent: Wed, Jun 15, 2011 3:11:00 Pm To: ccie_voice@onlinestudylist.com CC: Subject: [OSL | CCIE_Voice] how to verify class-based traffic shape? Per SRND, I configured FRF.12 as below: class-map match-any signal match dscp cs3 match dscp af31 class-map match-any voice match dscp ef ! policy-map wan-edge class voice priority percent 33 class signal bandwidth percent 5 class class-default fair-queue ! policy-map shape384 class class-default shape average 364800 3648 0 service-policy wan-edge ! interface Serial0/1/0.102 point-to-point bandwidth 384 frame-relay interface-dlci 102 class map384 What command we would use to verify the shaping? I tried command show traffic-shape but got nothing. show policy-map interface didn't give statistics on shaping. Thanks! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20110615/4c87f723/attachment-0001.html -- Message: 2 Date: Thu, 16 Jun 2011 14:11:36 +1000 From: Adil Shaikh adil.sha...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CUPC - hit and miss Message-ID: BANLkTi=HA=sX5gYF2DLhd7Zrp=ykdoh...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi All, Is there any gottcha for CUPC to see the presence of another phone when it goes off hook? I have done CUPS integration almost 6-7 times. Sometime the presence of another phone going off hook is displayed on CUPC and sometime not. If the CUPS integration is done today, then presence of contact in CUPC will be working perfectly fine for this integration. if tomorrow, i follow the same procedure for new integration for new lab, it will not work whether i reboot CUPS or not. The contact in CUPC is added from User Web Interface on CUPS. Has anyone come across similar
[OSL | CCIE_Voice] SRST and CUCM Route pattern conflict
Consider the Service provider expecting leading digit for international calls and NOT called party type. also you required to configure SRST on the GW which connected to PSTN and registered on CUCM by H323. When ever you dial international the leading digit should be send along with DNIS and 10 digit for ANI to the Service Provider. - on GW voice translation-rule 100 rule 1 /.*/ // voice translation-rule 10 rule 1 /^5...$/ /+1666222/ type any international plan any isdn voice translation-profile INT-Site-1 translate calling 10 translate called 100 dial-peer voice 100 pots translation-profile outgoing INT-Site-1 destination-pattern 9011T port 0/1/0:15 prefix 011 - On CUCM 1) Route List == Calling Party Transformation - Use Calling Party\'s Mask : ON - Calling Party Transformation Mask : +XX (10 digit) - Calling APrty Number Type : International - Calling Party Numbering Plan: ISDN == Called Party Transformation - Discard Digits : Predot - Called Party Transformation Mask : - Prefix Digits (Outgoing Calls) : 9011 - Called APrty Number Type : Cisco CallManager - Called Party Numbering Plan : Cisco CallManager 2) Route Pattern = 9011.! now this 10 digit calling number on CUCM not match with Voice translation rule 10 on GW and you never see 10 digit ANI on PSTN. To solve the issue: (either choices) Choice 1- need to use only 4 digit ANI on CUCM route pattern to match with translation-rule 10 on GW but you won't get mark because you did not configure 10 digit ANI on route pattern ! Choice 2- change the translation rule-10 to ten digit which you lose mark on SRST ! Any comment would be appreciate.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] SRST and CUCM Route pattern conflict
Hi, Basically for international calls you need to pass 10 digit ANI and leading number (011) along with DNIS to the Service Provider. Since your GW is H323, CUCM send the calls to GW then GW has to have such a dial-peer way out to PSTN. Midwhile your Dial-Peer has to match the requirement for your SRTS as well. Hopefully it is clear now. From: Roig Borrell, Francesc Xavier francesc.ro...@tecnocom.es To: Chris Green voice5...@yahoo.com; ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Sat, 11 June, 2011 13:22:37 Subject: RE: [OSL | CCIE_Voice] SRST and CUCM Route pattern conflict Hi Chris, I have read several times but I haven’t been able to understand the scenario. Could you clarify it a little bit? Thanks! De:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] En nombre de Chris Green Enviado el: sábado, 11 de junio de 2011 21:18 Para: ccie_voice@onlinestudylist.com Asunto: [OSL | CCIE_Voice] SRST and CUCM Route pattern conflict Consider the Service provider expecting leading digit for international calls and NOT called party type. also you required to configure SRST on the GW which connected to PSTN and registered on CUCM by H323. When ever you dial international the leading digit should be send along with DNIS and 10 digit for ANI to the Service Provider. - on GW voice translation-rule 100 rule 1 /.*/ // voice translation-rule 10 rule 1 /^5...$/ /+1666222/ type any international plan any isdn voice translation-profile INT-Site-1 translate calling 10 translate called 100 dial-peer voice 100 pots translation-profile outgoing INT-Site-1 destination-pattern 9011T port 0/1/0:15 prefix 011 - On CUCM 1) Route List == Calling Party Transformation - Use Calling Party\'s Mask : ON - Calling Party Transformation Mask : +XX (10 digit) - Calling APrty Number Type : International - Calling Party Numbering Plan: ISDN == Called Party Transformation - Discard Digits : Predot - Called Party Transformation Mask : - Prefix Digits (Outgoing Calls) : 9011 - Called APrty Number Type : Cisco CallManager - Called Party Numbering Plan : Cisco CallManager 2) Route Pattern = 9011.! now this 10 digit calling number on CUCM not match with Voice translation rule 10 on GW and you never see 10 digit ANI on PSTN. To solve the issue: (either choices) Choice 1- need to use only 4 digit ANI on CUCM route pattern to match with translation-rule 10 on GW but you won't get mark because you did not configure 10 digit ANI on route pattern ! Choice 2- change the translation rule-10 to ten digit which you lose mark on SRST ! Any comment would be appreciate.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] SRST and CUCM Route pattern conflict
Thank you George, I failed my exam and I got shock on Call routing mark which was too low, I am trying to dig down to it and find my mistakes. I debug isdn q931 and test the outputs one by one but I am not sure how I got very low mark on call routing. From: George Goglidze gogli...@gmail.com To: Chris Green voice5...@yahoo.com Cc: ccie_voice@onlinestudylist.com Sent: Sat, 11 June, 2011 13:09:28 Subject: Re: [OSL | CCIE_Voice] SRST and CUCM Route pattern conflict I'll reply under your choices... Choice 1- need to use only 4 digit ANI on CUCM route pattern to match with translation-rule 10 on GW but you won't get mark because you did not configure 10 digit ANI on route pattern ! Why do you think you will loose points? As long as provider sees 10digit ANI it should be fine, nobody will even check how you did it. they will just put debug and make a call probably. so you got your points here. Choice 2- change the translation rule-10 to ten digit which you lose mark on SRST ! this choice is valid too, but you must have two translations on the GW: voice translation-rule 10 rule 1 /^1666222/ /+/ type any international plan any isdn rule 2 /^5...$/ /+1666222/ type any international plan any isdn And you got your points here too... Hope this helps, On Sat, Jun 11, 2011 at 8:18 PM, Chris Green voice5...@yahoo.com wrote: Consider the Service provider expecting leading digit for international calls and NOT called party type. also you required to configure SRST on the GW which connected to PSTN and registered on CUCM by H323. When ever you dial international the leading digit should be send along with DNIS and 10 digit for ANI to the Service Provider. - on GW voice translation-rule 100 rule 1 /.*/ // voice translation-rule 10 rule 1 /^5...$/ /+1666222/ type any international plan any isdn voice translation-profile INT-Site-1 translate calling 10 translate called 100 dial-peer voice 100 pots translation-profile outgoing INT-Site-1 destination-pattern 9011T port 0/1/0:15 prefix 011 - On CUCM 1) Route List == Calling Party Transformation - Use Calling Party\'s Mask : ON - Calling Party Transformation Mask : +XX (10 digit) - Calling APrty Number Type : International - Calling Party Numbering Plan: ISDN == Called Party Transformation - Discard Digits : Predot - Called Party Transformation Mask : - Prefix Digits (Outgoing Calls) : 9011 - Called APrty Number Type : Cisco CallManager - Called Party Numbering Plan : Cisco CallManager 2) Route Pattern = 9011.! now this 10 digit calling number on CUCM not match with Voice translation rule 10 on GW and you never see 10 digit ANI on PSTN. To solve the issue: (either choices) Choice 1- need to use only 4 digit ANI on CUCM route pattern to match with translation-rule 10 on GW but you won't get mark because you did not configure 10 digit ANI on route pattern ! Choice 2- change the translation rule-10 to ten digit which you lose mark on SRST ! Any comment would be appreciate. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CME background with only 2 files
Extenok suggest the following: UCIP TelephonyCall ControlCUCMEConfig Examples Thank you so much for your suggestion as well. From: ccie_voice-requ...@onlinestudylist.com ccie_voice-requ...@onlinestudylist.com To: ccie_voice@onlinestudylist.com Sent: Thu, 9 June, 2011 9:00:02 Subject: CCIE_Voice Digest, Vol 64, Issue 90 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: CME background with only 2 files (Sam Park) 2. Re: pstn phone not getting dhcp address (George Goglidze) -- Message: 1 Date: Thu, 9 Jun 2011 11:07:51 -0400 From: Sam Park upperlevelpark...@gmail.com To: OSL ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME background with only 2 files Message-ID: BANLkTi==g2cnqx1hwbn1venx9c3mdft...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 The Link I use from the online doc is Voice and Video IP Telephony IP Phones 7900 Series New page Select: Maintain and Operate M O Guides 7965 for CUCM 7.0 (I think you can choose any version of CUCM) New page Select chapter: Customizing the Cisco Unified IP Phone This has the exact directory with numbers and formats you need, no need to remember. Cause, sometimes you panic and forget. This page also has the RingList.xml for ring tones. HTH Sam On Wed, Jun 8, 2011 at 3:09 PM, Extenok exte...@yahoo.com wrote: Hi, Another easier way is to go to the Cisco website and browse to: UCIP TelephonyCall ControlCUCMEConfig Examples Here you'll find a sample config for changing the background of a 7970 for a CME deployment, the sizing is same for a 7965 (320x212, i think). And the important thing, the List.xml format is the same for CUCM. Now you can focus on practicing something more complex 10 times a day :) E. Sent from my iPhone On Jun 8, 2011, at 10:07 AM, Victor Malyuga victor_maly...@yahoo.com wrote: Example of List.xml file is in the Phone Admin Guide in phone customisation section. The guide will be available in online documentation. --- On *Wed, 8/6/11, ShinGei Yong shingei.y...@gmail.com* wrote: From: ShinGei Yong shingei.y...@gmail.com Subject: Re: [OSL | CCIE_Voice] CME background with only 2 files To: Chris Green voice5...@yahoo.com Cc: ccie_voice@onlinestudylist.comccie_voice@onlinestudylist.com Date: Wednesday, 8 June, 2011, 4:18 You better to be familiar on how to write the List.xml script without referring to any doc during ur attempt. Not that too difficult, get yourself typing the below stuff 10times per day till the exam. CiscoIPPhoneImageList ImageItem Image=TFTP:Desktops/320x212x16/TN-Image.png URL=TFTP:Desktops/320x212x16/Image.png/ /CiscoIPPhoneImageList Shingei. On Wed, Jun 8, 2011 at 10:22 AM, Chris Green http://uk.mc290.mail.yahoo.com/mc/compose?to=voice5...@yahoo.com voice5...@yahoo.com wrote: Hi All, For CME Background we need 3 files, 2 images and one List XML file. Is there any way to complete this task with only with 2 images and not List XML file? Basically what would be the solution if you provided with only 2 images and NOT List XML file. Chris ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.com/www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out http://www.platinumplacement.com/www.PlatinumPlacement.com -Inline Attachment Follows- ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.comwww.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out http://www.PlatinumPlacement.comwww.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.comwww.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out http://www.PlatinumPlacement.comwww.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20110609/587cc30d/attachment-0001.html -- Message: 2
[OSL | CCIE_Voice] I just got my Fail result
Hi All, I just got my fail result, and what it shocked me is the Call Routing mark for 40% while I am sure I done what was asked for and I tested as well. I also got 100% in 4 different section but what it killed me is Call Routing which I was quite confident about that section :( Chris ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] digit manipulate on VOICE-PORT or on POT
Hi All, What is the difference manipulating the digit on VOICE-PORT or POT as follows? Which one expected for the exam and why? - voice translation-rule 1 rule 1 /^.*\(5...\)/ /\1/ voice translation-profile pstn-in translate called 1 voice-port 0/1/0:23 translation-profile incoming pstn-in dial-peer voice 1 pots incoming called-number . direct-inward-dial --- voice translation-rule 1 rule 1 /^.*\(5...\)/ /\1/ voice translation-profile pstn-in translate called 1 dial-peer voice 1 pots translation-profile incoming pstn-in incoming called-number . direct-inward-dial port 0/1/0:23 --- ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] No service configured on phone display
That is so complex specially if you have many phones, Consider you are working in a company with 2000 IP phones assigned to employees then every day you have request to run NO Service Configured for different employee who is getting time off and they need you block their directory for period of tie for security reason. Answer would be changing the External URL services... ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] I just failed the LAB 5
Hi Adam, Best of luck, you gonna make it this time, I got my fail result today but I have to find out my mistake and attend again. I am sure success marry you this time.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] CME background with only 2 files
Hi All, For CME Background we need 3 files, 2 images and one List XML file. Is there any way to complete this task with only with 2 images and not List XML file? Basically what would be the solution if you provided with only 2 images and NOT List XML file. Chris ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] GK-unreg-by-for
Hi All, I am using the dialplan-pattern 1 +1636303 extention-length 4 onSRST and isdn outgoing ie redirecting-number on Serial0/1/0:23 R2-phone is unregistered on CUCM and it is on SRST mode R3-phone is registered in CME and need to call R2-phone through GateKeeper in normal case as well as SRST When R2-phone goes to SRST mode I am not able to make the call from R3-phone through GK-CCM and also I can not see by and for when I am calling R2-phone in SRST mode Kindly let me know what am I missing ? Regards Chris ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 63, Issue 176
add / at directory field From: ccie_voice-requ...@onlinestudylist.com ccie_voice-requ...@onlinestudylist.com To: ccie_voice@onlinestudylist.com Sent: Thu, 26 May, 2011 18:57:37 Subject: CCIE_Voice Digest, Vol 63, Issue 176 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: Help (Vik Malhi) 2. RingList.xml not updating?! (Ki Wi) 3. Re: RingList.xml not updating?! (Ki Wi) 4. Re: RingList.xml not updating?! (Cristobal Priego) 5. Re: RingList.xml not updating?! (George Goglidze) -- Message: 1 Date: Thu, 26 May 2011 13:12:27 -0700 From: Vik Malhi vma...@ipexpert.com To: Bill Lake whl...@gmail.com,OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Help Message-ID: ca0401a2.11931%vma...@ipexpert.com Content-Type: text/plain;charset=ISO-8859-1 If this has been resolved- apologies for the redundant info. The bearer capability for the failed call is different. Bearer Capability i = 0x8890 Standard = CCITT Transfer Capability = Unrestricted Digital 0x8890 is 64k data call. Try this command: SJC-RTR(config)#voice-port 0/0/0:23 SJC-RTR(config-voiceport)#bearer-cap speech -- Vik Malhi ? CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 ext 420 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, Wireless, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ On 5/25/11 5:35 PM, Bill Lake whl...@gmail.com wrote: Hello Everyone, I ran into a real world issue that I am having trouble with. We upgraded a 10 b channel PRI to a new PRI that has 23 b channels and afterwards some of the IP phones can not make outbound calls. These phones all have a shared main line and 3 of phones cannot make outbound calls on that line while all the others can. I have changed the numbers and minimized the names to protect the innocent but has anyone seen anything like this. The dial peer is matched the same and below is the debug isdn q931 and as you can see, to calls go through fine but one of the phones with the issue has the error Cause i = 0x83A9 - Temporary failure It does not make sense since the dial peer match is the same and as you can see below it gets sent to the ISDN circuit but then disconnects. Another odd thing is one phone fast busy's once and disconnects while the other just get fast busy. This is a local h323 gateway configured on the CUCM but no gatekeeper or anything else that I am aware of. All phones are 7962. Any help would be greatly appriciated. *May 25 2011 16:47:45.363 CDT: ISDN Se0/0/1:23 Q931: pak_private_number: Invalid type/plan 0x0 0x0 may be overriden; sw-type 13 *May 25 2011 16:47:45.363 CDT: ISDN Se0/0/1:23 Q931: pak_private_number: Invalid type/plan 0x0 0x0 may be overriden; sw-type 13 *May 25 2011 16:47:45.363 CDT: ISDN Se0/0/1:23 Q931: Sending SETUP callref = 0x00B1 callID = 0x8032 switch = primary-ni interface = User *May 25 2011 16:47:45.367 CDT: ISDN Se0/0/1:23 Q931: TX - SETUP pd = 8 callref = 0x00B1 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Facility i = 0x9F8B0100A11F0201330201008017427265616B20526D202D20434845535445524649454C 44 Protocol Profile = Networking Extensions 0xA11F0201330201008017427265616B20526D202D20434845535445524649454C44 Component = Invoke component Invoke Id = 51 Operation = CallingName Name Presentation Allowed Extended Name = Break Rm Display i = 'Break Rm' Calling Party Number i = 0x2181, '6365551234' Plan:ISDN, Type:National Called Party Number i = 0xA1,
