Re: [OSL | CCIE_Voice] New Lab Release
I went for the lab, there is no lab 5 From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of rsvpaar Sent: Sunday, June 05, 2011 3:25 PM To: John Persi ; ccie_voice Subject: Re: [OSL | CCIE_Voice] New Lab Release Hi John, Do u have latest lab i am ready to share the cost and discuss i saw it was same to same i got in my lab ! On Sun, 05 Jun 2011 16:28:37 +0530 John Persi john_pe...@yahoo.commailto:john_pe...@yahoo.com wrote Kindly PM me if anyone ready to discuss Thanks From: John Persi john_pe...@yahoo.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Sunday, June 5, 2011 2:49 PM Subject: New Lab Release Hi, Does anyone ready to discussnew lab. I can see on cert knowledge . com but it is expensive http:// www . certknowledge . com/forum/index.php?topic=39.15http://%20www%20. %20certknowledge %20 . %20com/forum/index.php?topic=39.15 No budget already attempted for 3 times. Regards Jonny ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.PlatinumPlacement.com [http://sigads.rediff.com/RealMedia/ads/adstream_nx.ads/www.rediffmail.com/signatureline.htm@Middle]http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm@Middle? Treat yourself at a restaurant, spa, resort and much more with Rediff Deal ho jaye!http://track.rediff.com/click?url=___http://dealhojaye.rediff.com/___cmp=signaturelnk=rediffmailsignaturenewservice=deals * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ 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] Gatekeeper question
Go to service parameters, choose call manager and search for trunk and will find the option to make the gatekeeper h225 trunk to use 1720, and then put the name of your h225 trunk which is GK-TRUNK From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Randall Crumm Sent: Monday, May 30, 2011 9:39 PM To: Online Study Subject: [OSL | CCIE_Voice] Gatekeeper question Hi, How do I get gk-trunk_1 and 2 to repflect port 1720? Thanks, Randall HQ-RTR(config)#do show gatekeeper gw GATEWAY TYPE PREFIX TABLE = Prefix: 3* Zone HQ master gateway list: 10.10.110.3:1720 BR2-RTR Prefix: 1#* Zone HQ master gateway list: 10.10.210.10:40081 GK-TRUNK_1 10.10.210.11:33277 GK-TRUNK_2 Zone HQ prefix 1... priority gateway list(s): Priority 5: 10.10.210.10:40081 GK-TRUNK_1 10.10.210.11:33277 GK-TRUNK_2 Zone HQ prefix 5... priority gateway list(s): Priority 5: 10.10.210.10:40081 GK-TRUNK_1 10.10.210.11:33277 GK-TRUNK_2 * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ 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] Finally, I pass the LAB :D
Hi Roberto, I hope you're doing fine. Can you please share your experience with me? I'm going for the exam very soon. I'm really ready but I need some trick from you to complete this CCIE :) Waiting for your feedback Thanks Naoufal -Original Message- From: Naoufal Kerboute Sent: Wednesday, May 18, 2011 2:32 PM To: 'Roberto Reyes Alanis'; ccie_voice@onlinestudylist.com Subject: RE: Finally, I pass the LAB :D CONGRATULATIONS MAN GREAT JOB :D Naoufal -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roberto Reyes Alanis Sent: Wednesday, May 18, 2011 10:54 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Finally, I pass the LAB :D I only want say Yahooo and Tks to Naoufal Kerboute. Y para los de America Latina salu2 y todo pa delante. Roberto Reyes Alanis CCIE#28945. ___ 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 * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ 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] OT: UCCX Script, Virtual Queueing
Hi, Check the link below, and search for the script (callkback). http://uccx.net/uccx-7x-sample-scripts.html Naoufal From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of George Goglidze Sent: Friday, May 20, 2011 6:01 PM To: CCIE Voice online groupstudy Subject: [OSL | CCIE_Voice] OT: UCCX Script, Virtual Queueing Hi all, I do appologize for this OT, but I searched the whole google and big part of cisco.comhttp://cisco.com and couldn't find the answer to my question. One of our customers have the following requirement: when a caller calls, if he is in a queue too long (configured threshold), he will be offered an option to leave his number and the system will virtually keep him in a queue, and when his time comes, and agent is available, the system will call the user back and connect him to the agent. ok, now I've given it a thought, and there is no way you can be in a queue on UCCX, if there is no Active Contact in a script. Basically just to enter the Select Resrouce step, you must specify a contact that is Active otherwise it does not work. I had an idea, which I understand is a very bad design, and very bad usage of UCCX Resources, but here it is: I could then place a call from the script into UCCX again, and create this Active contact myself. and then place this Active Contact into a queue, but I don't like this design because it's using double of resources of UCCX. Before providing this solution to my customer, I would like to ask if anyone has done anything like this, and if you know a better solution of doing this. Many thanks, and sorry for OT again, but I couldn't think of a better place to ask this. Regards, * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ 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] DUBAI EXAM
Hi Guys, I feel that no one is passing the exam in Dubai, what do you think? Thanks Naoufal * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ 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] Finally, I pass the LAB :D
CONGRATULATIONS MAN GREAT JOB :D Naoufal -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roberto Reyes Alanis Sent: Wednesday, May 18, 2011 10:54 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Finally, I pass the LAB :D I only want say Yahooo and Tks to Naoufal Kerboute. Y para los de America Latina salu2 y todo pa delante. Roberto Reyes Alanis CCIE#28945. ___ 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 * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ 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] DUBAI EXAM
When? From: Shady Hasan [mailto:shady@gmail.com] Sent: Wednesday, May 18, 2011 3:03 PM To: Naoufal Kerboute Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] DUBAI EXAM I know 2 CCIEs Voice passed from Dubai :) On Wed, May 18, 2011 at 12:14 PM, Naoufal Kerboute naou...@mhdinfotech.commailto:naou...@mhdinfotech.com wrote: Hi Guys, I feel that no one is passing the exam in Dubai, what do you think? Thanks Naoufal * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.PlatinumPlacement.com * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ 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] VoiceMail (Exam result FAILED)
Hi guys, I failed, and I got 19% in voicemail, and I was sure that everything is ok, mailbox created with the password 12345 (as asked in the exam), forward to VM after 20s for all phones, MWI was working I've checked that many times. For CUE the same thing MWI and mailbox working, What could be wrong? Until now I'm sure that I've tested voicemail (MWI, mailbox, password), users imported from CUCM as requested. Any ideas? Naoufal * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ 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] Spoken Name Script
Hi guys, I found the below link and I want to do the same, it's very interesting and this is what I'm looking for exactly, but Im facing an issue and I need your help. This is the script: select resource selected play prompt selected resources id connect connected end failed goto start_of_queue queued label start_of_queue ... The problem is in the 1st Select_resource I stopped the call to connect to save the userID in a variable user, but I want to play this variable which is user but I can't. My issue is how to convert this User variable which contain the spoken Name to a prompt and then play the same before my 2nd connect. Any Idea? This is really interesting scenario and we could have it in the exam, believe me. Waiting for your feedback. PS: I'm using the default spoken Name script to record the names. Thanks a lot Naoufal From: George Goglidze [mailto:gogli...@gmail.com] Sent: Tuesday, May 03, 2011 1:38 PM To: Naoufal Kerboute Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Spoken Name Script Hi, To use the spoken name in another script, you need two steps, at least: 1) Get User, where you can provide username, or agent extension to get the user info. 2) Get User Info, here you can retrieve Spoken Name into a prompt variable. I hope this helps, Regards, On Mon, May 2, 2011 at 8:00 PM, Naoufal Kerboute naou...@mhdinfotech.commailto:naou...@mhdinfotech.com wrote: OK, how can I use the spoken name in another script, how to play the name of agent when routing the call to it ? From: George Goglidze [mailto:gogli...@gmail.commailto:gogli...@gmail.com] Sent: Monday, May 02, 2011 11:01 PM To: Naoufal Kerboute Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Spoken Name Script Hi, Spoken name script's sole function is to record agent's name... Then from any script, if you need agent's name to be played, you can play this back. or otherwise you can generate prompt with agent's name and surname values and play it back. but normally recorded prompt gives better result. Regards, On Mon, May 2, 2011 at 7:17 PM, Naoufal Kerboute naou...@mhdinfotech.commailto:naou...@mhdinfotech.com wrote: Hi guys, I have some confusion regarding the spoken name script. Is there anyone who can help me to understand it? What I can do with it? How I can use it? What is the goal of this script? Thanks Naoufal NAOUFAL KERBOUTE TECHNICAL MANAGER Post Box 880, Postal Code 112, Ruwi Sultanate of Oman Telephone 24 835 252, 24 834 848 24 834 400, 24 836 226 Mobile 9604 2593 E-Mail naou...@mhdinfotech.commailto:naou...@mhdinfotech.com Web www.mhdinfotech.comhttp://www.mhdinfotech.com/ * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.PlatinumPlacement.com * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system
Re: [OSL | CCIE_Voice] Spoken Name Script
I forgot to attach the link https://supportforums.cisco.com/message/3306933#3306933 From: Naoufal Kerboute Sent: Thursday, May 05, 2011 3:58 PM To: 'George Goglidze'; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Spoken Name Script Hi guys, I found the below link and I want to do the same, it's very interesting and this is what I'm looking for exactly, but Im facing an issue and I need your help. This is the script: select resource selected play prompt selected resources id connect connected end failed goto start_of_queue queued label start_of_queue ... The problem is in the 1st Select_resource I stopped the call to connect to save the userID in a variable user, but I want to play this variable which is user but I can't. My issue is how to convert this User variable which contain the spoken Name to a prompt and then play the same before my 2nd connect. Any Idea? This is really interesting scenario and we could have it in the exam, believe me. Waiting for your feedback. PS: I'm using the default spoken Name script to record the names. Thanks a lot Naoufal From: George Goglidze [mailto:gogli...@gmail.com] Sent: Tuesday, May 03, 2011 1:38 PM To: Naoufal Kerboute Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Spoken Name Script Hi, To use the spoken name in another script, you need two steps, at least: 1) Get User, where you can provide username, or agent extension to get the user info. 2) Get User Info, here you can retrieve Spoken Name into a prompt variable. I hope this helps, Regards, On Mon, May 2, 2011 at 8:00 PM, Naoufal Kerboute naou...@mhdinfotech.commailto:naou...@mhdinfotech.com wrote: OK, how can I use the spoken name in another script, how to play the name of agent when routing the call to it ? From: George Goglidze [mailto:gogli...@gmail.commailto:gogli...@gmail.com] Sent: Monday, May 02, 2011 11:01 PM To: Naoufal Kerboute Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Spoken Name Script Hi, Spoken name script's sole function is to record agent's name... Then from any script, if you need agent's name to be played, you can play this back. or otherwise you can generate prompt with agent's name and surname values and play it back. but normally recorded prompt gives better result. Regards, On Mon, May 2, 2011 at 7:17 PM, Naoufal Kerboute naou...@mhdinfotech.commailto:naou...@mhdinfotech.com wrote: Hi guys, I have some confusion regarding the spoken name script. Is there anyone who can help me to understand it? What I can do with it? How I can use it? What is the goal of this script? Thanks Naoufal NAOUFAL KERBOUTE TECHNICAL MANAGER Post Box 880, Postal Code 112, Ruwi Sultanate of Oman Telephone 24 835 252, 24 834 848 24 834 400, 24 836 226 Mobile 9604 2593 E-Mail naou...@mhdinfotech.commailto:naou...@mhdinfotech.com Web www.mhdinfotech.comhttp://www.mhdinfotech.com/ * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.PlatinumPlacement.com * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use
Re: [OSL | CCIE_Voice] Spoken Name Script
Ignore it, I'm done :) Thanks George, without your help I'll never fix it. Naoufal From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Naoufal Kerboute Sent: Thursday, May 05, 2011 3:53 PM To: George Goglidze; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Spoken Name Script Hi guys, I found the below link and I want to do the same, it's very interesting and this is what I'm looking for exactly, but Im facing an issue and I need your help. This is the script: select resource selected play prompt selected resources id connect connected end failed goto start_of_queue queued label start_of_queue ... The problem is in the 1st Select_resource I stopped the call to connect to save the userID in a variable user, but I want to play this variable which is user but I can't. My issue is how to convert this User variable which contain the spoken Name to a prompt and then play the same before my 2nd connect. Any Idea? This is really interesting scenario and we could have it in the exam, believe me. Waiting for your feedback. PS: I'm using the default spoken Name script to record the names. Thanks a lot Naoufal From: George Goglidze [mailto:gogli...@gmail.com] Sent: Tuesday, May 03, 2011 1:38 PM To: Naoufal Kerboute Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Spoken Name Script Hi, To use the spoken name in another script, you need two steps, at least: 1) Get User, where you can provide username, or agent extension to get the user info. 2) Get User Info, here you can retrieve Spoken Name into a prompt variable. I hope this helps, Regards, On Mon, May 2, 2011 at 8:00 PM, Naoufal Kerboute naou...@mhdinfotech.commailto:naou...@mhdinfotech.com wrote: OK, how can I use the spoken name in another script, how to play the name of agent when routing the call to it ? From: George Goglidze [mailto:gogli...@gmail.commailto:gogli...@gmail.com] Sent: Monday, May 02, 2011 11:01 PM To: Naoufal Kerboute Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Spoken Name Script Hi, Spoken name script's sole function is to record agent's name... Then from any script, if you need agent's name to be played, you can play this back. or otherwise you can generate prompt with agent's name and surname values and play it back. but normally recorded prompt gives better result. Regards, On Mon, May 2, 2011 at 7:17 PM, Naoufal Kerboute naou...@mhdinfotech.commailto:naou...@mhdinfotech.com wrote: Hi guys, I have some confusion regarding the spoken name script. Is there anyone who can help me to understand it? What I can do with it? How I can use it? What is the goal of this script? Thanks Naoufal NAOUFAL KERBOUTE TECHNICAL MANAGER Post Box 880, Postal Code 112, Ruwi Sultanate of Oman Telephone 24 835 252, 24 834 848 24 834 400, 24 836 226 Mobile 9604 2593 E-Mail naou...@mhdinfotech.commailto:naou...@mhdinfotech.com Web www.mhdinfotech.comhttp://www.mhdinfotech.com/ * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.PlatinumPlacement.com * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified
[OSL | CCIE_Voice] Remote Gatekeeper Troubleshooting
Hi guys, I feel that this mailing list is sleeping :) Any way I have a question and I hope that I get an answer from you guys. I'm playing with gatekeeper and I'm thinking of different way to troubleshoot the connection between my local gatekeeper and a remote one (I don't have access to remote GK). What is the best way to troubleshoot a broken connection between two gatekeeper? How can I check from debugs which prefix or default technology prefix the remote GK is using, or the bandwidth defined in the remote GK, let assume that the remote GK enable only g729 and I'm sending the call with g711, from the debug I'll see BRJ_INSUFFICIENT_RSC but how can I know which bandwidth is allocated for me so I can define a proper codec information. Waiting for your feedbacks Thanks a lot Naoufal * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ 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] Finally succeeded ..Got CCIE
Congratulations man. You're the BOSS now :) From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Shrini Sent: Wednesday, May 04, 2011 12:02 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Finally succeeded ..Got CCIE Hi Experts, Today I am officially announced as Voice CCIE. Thanks to one and all for your valuable suggestions and help throughout this journey. Special thanks to Vik and IP Expert team for hosting this excellent mailer list and helping us. Thanks again Shrini * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Remote Gatekeeper Troubleshooting
No reply :s From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Naoufal Kerboute Sent: Wednesday, May 04, 2011 1:01 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Remote Gatekeeper Troubleshooting Hi guys, I feel that this mailing list is sleeping :) Any way I have a question and I hope that I get an answer from you guys. I'm playing with gatekeeper and I'm thinking of different way to troubleshoot the connection between my local gatekeeper and a remote one (I don't have access to remote GK). What is the best way to troubleshoot a broken connection between two gatekeeper? How can I check from debugs which prefix or default technology prefix the remote GK is using, or the bandwidth defined in the remote GK, let assume that the remote GK enable only g729 and I'm sending the call with g711, from the debug I'll see BRJ_INSUFFICIENT_RSC but how can I know which bandwidth is allocated for me so I can define a proper codec information. Waiting for your feedbacks Thanks a lot Naoufal * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ 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] Spoken Name Script
Hi, Where the spoken name script save the wav files? Also do you have any idea how to use play the name of the agent before routing the call to it? Thank a lot Naoufal From: George Goglidze [mailto:gogli...@gmail.com] Sent: Tuesday, May 03, 2011 1:38 PM To: Naoufal Kerboute Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Spoken Name Script Hi, To use the spoken name in another script, you need two steps, at least: 1) Get User, where you can provide username, or agent extension to get the user info. 2) Get User Info, here you can retrieve Spoken Name into a prompt variable. I hope this helps, Regards, On Mon, May 2, 2011 at 8:00 PM, Naoufal Kerboute naou...@mhdinfotech.commailto:naou...@mhdinfotech.com wrote: OK, how can I use the spoken name in another script, how to play the name of agent when routing the call to it ? From: George Goglidze [mailto:gogli...@gmail.commailto:gogli...@gmail.com] Sent: Monday, May 02, 2011 11:01 PM To: Naoufal Kerboute Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Spoken Name Script Hi, Spoken name script's sole function is to record agent's name... Then from any script, if you need agent's name to be played, you can play this back. or otherwise you can generate prompt with agent's name and surname values and play it back. but normally recorded prompt gives better result. Regards, On Mon, May 2, 2011 at 7:17 PM, Naoufal Kerboute naou...@mhdinfotech.commailto:naou...@mhdinfotech.com wrote: Hi guys, I have some confusion regarding the spoken name script. Is there anyone who can help me to understand it? What I can do with it? How I can use it? What is the goal of this script? Thanks Naoufal NAOUFAL KERBOUTE TECHNICAL MANAGER Post Box 880, Postal Code 112, Ruwi Sultanate of Oman Telephone 24 835 252, 24 834 848 24 834 400, 24 836 226 Mobile 9604 2593 E-Mail naou...@mhdinfotech.commailto:naou...@mhdinfotech.com Web www.mhdinfotech.comhttp://www.mhdinfotech.com/ * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.PlatinumPlacement.com * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you
[OSL | CCIE_Voice] Calling Number (Type and plan)
Hi guys, I need an advice from CCIE's who already passed the exam. In the call routing section, if the question doesn't ask for the calling number type and plan, do I have to set the calling number type (Sub, Nat ...) vs the called number or no need? Also if the question doesn't ask for calling name do I have to restricted? Thanks Naoufal * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ 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 busy-trigger-button Problems
The max calls per button for outgoing and incoming calls From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Shrini Sent: Tuesday, May 03, 2011 9:33 PM To: Roig Borrell, Francesc Xavier Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME busy-trigger-button Problems Actually max calls per button is for outgoing calls. Below one I configured a while ago, will try later today. Thanks Shrini On 5/3/2011 10:02 AM, Shrini wrote: Hi Roig, Try adding max-calls-per-button 5 Thanks Shrini On 5/3/2011 9:23 AM, Roig Borrell, Francesc Xavier wrote: Hi Shrini, Sorry for not answering you before. I had to stop my studies for a while due to work :-( Now I have tested it and here are my conclusions Ephone-dn 11 octo-line Number 4023 Huntstop channel 5 ephone 1 busy-trigger-per-button 4 button 1:1 2:11 ! ephone 2 busy-trigger-per-button 2 button 1:2 2:11 1st call answered by phone2 2nd call answered by phone2 3rd call answered by phone 1 4th call answered by phone 1 5th call busy! The first two calls answered by phone 2 occupy a 2 channels of shared dn 11 in phone 1 . So the busy trigger of phone 1 (busy trigger 4) will allow only 2 more incoming calls. Looking at the sh ephone before making the 5h call, 4 channels are occupied for phone 1. So the next call will always give busy. Have you tested? Does it work for you? Or am I missing anything and there is a way to achieve this? Thanks ephone-1[0] Mac:0024.97AA.1B72 TCP socket:[3] activeLine:2 REGISTERED in SCCP ver 17/9 mediaActive:1 offhook:1 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:9 privacy:1 IP:192.168.22.50 38312 7945 keepalive 16 max_line 2 button 1: dn 1 number 4001 CH1 IDLE CH2 IDLE CH3 IDLE CH4 IDLE CH5 IDLE CH6 IDLE CH7 IDLE CH8 IDLE button 2: dn 11 number 4023 CH1 HOLD CH2 CONNECTEDCH3 HOLD CH4 CONNECTEDCH5 IDLE CH6 IDLE CH7 IDLE CH8 IDLE shared paging-dn 9 Preferred Codec: g711ulaw Active Call on DN 11 chan 4 :4023 192.168.22.50 20938 to 142.102.66.254 2000 via 10.10.112.1 G729 20 bytes no vad Tx Pkts 444 bytes 14208 Rx Pkts 228 bytes 7296 Lost 218 Jitter 78 Latency 0 callingDn -1 calledDn -1 4 calls are visible on line 2 Username: br2ph1 ephone-2[1] Mac:0024.97AA.1B49 TCP socket:[2] activeLine:2 REGISTERED in SCCP ver 17/9 mediaActive:1 offhook:1 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:9 privacy:1 IP:192.168.22.51 45034 7945 keepalive 15 max_line 2 button 1: dn 2 number 4002 CH1 IDLE CH2 IDLE CH3 IDLE CH4 IDLE CH5 IDLE CH6 IDLE CH7 IDLE CH8 IDLE button 2: dn 11 number 4023 CH1 HOLD CH2 CONNECTEDCH3 HOLD CH4 CONNECTEDCH5 IDLE CH6 IDLE CH7 IDLE CH8 IDLE shared paging-dn 9 Preferred Codec: g711ulaw Active Call on DN 11 chan 2 :4023 192.168.22.51 29502 to 142.102.66.254 2000 via 10.10.112.1 G729 20 bytes no vad Tx Pkts 1178 bytes 37696 Rx Pkts 295 bytes 9440 Lost 812 Jitter 158 Latency 0 callingDn -1 calledDn -1 4 calls are visible on line 2 Username: br2ph2 De: Shrini [mailto:linuxbos...@gmail.com] Enviado el: sábado, 23 de abril de 2011 2:09 Para: Roig Borrell, Francesc Xavier; 'Peter Farkas'; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Asunto: RE: [OSL | CCIE_Voice] CME busy-trigger-button Problems Roig, This will do , lets assume you made 5 calls to the dn. ephone-dn 10 octo number 1000 huntstop channel 5 This will limit 5 calls ephone 1 busy-trigger-per-button 4 -- 3rd 4th and 5th call you can accept here button 1:10 ephone 2 busy-trigger-per-button 2 --- 1st and 2nd call ringed here and picked here (3rd call will not ring here) button 1:10 You can test vice versa. Thanks Shrini From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roig Borrell, Francesc Xavier Sent: Friday, April 22, 2011 5:45 AM To: Peter Farkas; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME busy-trigger-button Problems Hi Peter, You are right. Testing it, the busy trigger applies in the shared line So how could we achieve this? shared line -1) Maximum 5 incoming calls into the DN -2) 1st phone should not receive not more than 4 incoming call and 2nd phone Should not receive more than two I have seen several posts proposing this config. But testing in the lab it doesn't work for me Due to previous reason (busy-trigger-per-button 4) the fifth call always hears busy tone http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg19976.html ephone-dn 10 octo number 1000 huntstop
[OSL | CCIE_Voice] SRST (name = EPNM)
Hi guys, Is there any way to force phones to not take name as EPNM in SRST mode (Call-manager-fallback)? Thanks Naoufal * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ 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] Spoken Name Script
Hi guys, I have some confusion regarding the spoken name script. Is there anyone who can help me to understand it? What I can do with it? How I can use it? What is the goal of this script? Thanks Naoufal NAOUFAL KERBOUTE TECHNICAL MANAGER [Description: Description: MHD_infotech] Post Box 880, Postal Code 112, Ruwi Sultanate of Oman Telephone 24 835 252, 24 834 848 24 834 400, 24 836 226 Mobile 9604 2593 E-Mail naou...@mhdinfotech.commailto:naou...@mhdinfotech.com Web www.mhdinfotech.comhttp://www.mhdinfotech.com/ * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * inline: image001.png___ 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] Spoken Name Script
OK, how can I use the spoken name in another script, how to play the name of agent when routing the call to it ? From: George Goglidze [mailto:gogli...@gmail.com] Sent: Monday, May 02, 2011 11:01 PM To: Naoufal Kerboute Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Spoken Name Script Hi, Spoken name script's sole function is to record agent's name... Then from any script, if you need agent's name to be played, you can play this back. or otherwise you can generate prompt with agent's name and surname values and play it back. but normally recorded prompt gives better result. Regards, On Mon, May 2, 2011 at 7:17 PM, Naoufal Kerboute naou...@mhdinfotech.commailto:naou...@mhdinfotech.com wrote: Hi guys, I have some confusion regarding the spoken name script. Is there anyone who can help me to understand it? What I can do with it? How I can use it? What is the goal of this script? Thanks Naoufal NAOUFAL KERBOUTE TECHNICAL MANAGER [Description: Description: MHD_infotech] Post Box 880, Postal Code 112, Ruwi Sultanate of Oman Telephone 24 835 252, 24 834 848 24 834 400, 24 836 226 Mobile 9604 2593 E-Mail naou...@mhdinfotech.commailto:naou...@mhdinfotech.com Web www.mhdinfotech.comhttp://www.mhdinfotech.com/ * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.PlatinumPlacement.com * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ 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] Spoken Name Script
Also please how to enter the username in the keyboard? Right now I'm just using Q=7 and Z=9 and the Name dialing of this user is = qz From: George Goglidze [mailto:gogli...@gmail.com] Sent: Monday, May 02, 2011 11:01 PM To: Naoufal Kerboute Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Spoken Name Script Hi, Spoken name script's sole function is to record agent's name... Then from any script, if you need agent's name to be played, you can play this back. or otherwise you can generate prompt with agent's name and surname values and play it back. but normally recorded prompt gives better result. Regards, On Mon, May 2, 2011 at 7:17 PM, Naoufal Kerboute naou...@mhdinfotech.commailto:naou...@mhdinfotech.com wrote: Hi guys, I have some confusion regarding the spoken name script. Is there anyone who can help me to understand it? What I can do with it? How I can use it? What is the goal of this script? Thanks Naoufal NAOUFAL KERBOUTE TECHNICAL MANAGER [Description: Description: MHD_infotech] Post Box 880, Postal Code 112, Ruwi Sultanate of Oman Telephone 24 835 252, 24 834 848 24 834 400, 24 836 226 Mobile 9604 2593 E-Mail naou...@mhdinfotech.commailto:naou...@mhdinfotech.com Web www.mhdinfotech.comhttp://www.mhdinfotech.com/ * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.PlatinumPlacement.com * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ 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] Workbook 2 Lab 2 Task 6.3
Yes guys you're right, the 10 50 50 20 couldn't be the shape value From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Steve Denney (stdenney) Sent: Monday, May 02, 2011 10:38 PM To: Pablo Meneses Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Workbook 2 Lab 2 Task 6.3 You know, I think you're right. The 10 50 20 20 would seem to be correct for *share* not shape (since 50 / (10+50+20+20) = 50/100 = 1/2 = 50%). Anyone else? cheers, sd From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Pablo Meneses Sent: Saturday, April 30, 2011 7:46 PM To: ccie voice Subject: [OSL | CCIE_Voice] Workbook 2 Lab 2 Task 6.3 Hello Experts, I was wondering if one of you could give me an explanation on this task since I am a bit lost: Workbook 2 Lab 2 Task 6.3: The task says: Ensure that the SRR scheduler for phones at the HQ switch shapes Q2 to 50% of the interface bandwidth. On the Solution Guide, it says: HQ-3750(config)#interface FastEthernet 1/0/2 HQ-3750(config-if)#srr-queue bandwidth shape 10 50 20 20 After I did that I got the following output: HQ-3750#show mls qos interface fastEthernet 1/0/2 queueing FastEthernet1/0/2 Egress Priority Queue : enabled Shaped queue weights (absolute) : 10 50 20 20 Shared queue weights : 10 10 60 20 The port bandwidth limit : 100 (Operational Bandwidth:100.0) The port is mapped to qset : 2 However, I then check CCO and found the following: The bandwidth weight for queue 1 is 1/8, which is 12.5 percent: Switch(config)# interface gigabitethernet2/0/1 Switch(config-if)# srr-queue bandwidth shape 8 0 0 0 http://www.cisco.com/en/US/docs/switches/lan/catalyst3750/software/release/12.2_50_se/configuration/guide/swqos.html#wp1163879 The question is: Why is it configured on the solution as 10 50 20 20? Shouldn't it be configured as 0 2 0 0 since 1/2 equals 0.5? Looking forward to your response. -Pablo Meneses. * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ 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] Workbook 2 Lab 2 Task 6.3
Hi, Below the solution interface FastEthernet1/0/2 srr-queue bandwidth share 10 10 60 20 priority-queue out Q1 is configured for the PQ. We will set the shape for Q2 to be 50% (guarantee and rate limit traffic in Q2 to 50%). To do this we specify a value of 2 for Q2 = ½= 50%. HQ-3750(config)#interface FastEthernet1/0/2 HQ-3750(config-if)#srr-queue bandwidth shape 0 2 0 0 VOILA HQ-3750#sh mls qos interface F1/0/2 queueing FastEthernet1/0/2 Egress Priority Queue : enabled Shaped queue weights (absolute) : 0 2 0 0 Shared queue weights : 10 10 60 20 The port bandwidth limit : 100 (Operational Bandwidth:100.0) The port is mapped to qset : 1 HQ-3750(config)#interface FastEthernet1/0/2 HQ-3750(config-if)# queue-set 2 Rgds Naoufal From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Steve Denney (stdenney) Sent: Monday, May 02, 2011 10:38 PM To: Pablo Meneses Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Workbook 2 Lab 2 Task 6.3 You know, I think you're right. The 10 50 20 20 would seem to be correct for *share* not shape (since 50 / (10+50+20+20) = 50/100 = 1/2 = 50%). Anyone else? cheers, sd From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Pablo Meneses Sent: Saturday, April 30, 2011 7:46 PM To: ccie voice Subject: [OSL | CCIE_Voice] Workbook 2 Lab 2 Task 6.3 Hello Experts, I was wondering if one of you could give me an explanation on this task since I am a bit lost: Workbook 2 Lab 2 Task 6.3: The task says: Ensure that the SRR scheduler for phones at the HQ switch shapes Q2 to 50% of the interface bandwidth. On the Solution Guide, it says: HQ-3750(config)#interface FastEthernet 1/0/2 HQ-3750(config-if)#srr-queue bandwidth shape 10 50 20 20 After I did that I got the following output: HQ-3750#show mls qos interface fastEthernet 1/0/2 queueing FastEthernet1/0/2 Egress Priority Queue : enabled Shaped queue weights (absolute) : 10 50 20 20 Shared queue weights : 10 10 60 20 The port bandwidth limit : 100 (Operational Bandwidth:100.0) The port is mapped to qset : 2 However, I then check CCO and found the following: The bandwidth weight for queue 1 is 1/8, which is 12.5 percent: Switch(config)# interface gigabitethernet2/0/1 Switch(config-if)# srr-queue bandwidth shape 8 0 0 0 http://www.cisco.com/en/US/docs/switches/lan/catalyst3750/software/release/12.2_50_se/configuration/guide/swqos.html#wp1163879 The question is: Why is it configured on the solution as 10 50 20 20? Shouldn't it be configured as 0 2 0 0 since 1/2 equals 0.5? Looking forward to your response. -Pablo Meneses. * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ 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] MGCP BR1 registration problem
Check the framing and the linecode should be the same as your PSTN router. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ahmed Ellboudy Sent: Monday, May 02, 2011 5:05 AM To: CCIE Voice Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MGCP BR1 registration problem Still did not solve it . Any idea? Thanks, Ahmed Ellboudy | CCNP, CCVP. Networking Team Leader Raya IT - Professional Networking Services Mobile: +20100770837 Tel : +20238276000 Ext. 2338 Fax : +20238372930 Email : ahmed_ellbo...@rayacorp.commailto:nadia_khal...@rayacorp.com Address : El Motamayez District - 6th of October [cid:image001.jpg@01CB8A26.89E6B660] From: CCIE Voice [mailto:cc...@corb.net] Sent: Monday, May 02, 2011 2:34 AM To: Ahmed Ellboudy Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MGCP BR1 registeration problem Try reloading the router. -- On May 1, 2011, at 17:43, Ahmed Ellboudy ahmed_ellbo...@rayacorp.commailto:ahmed_ellbo...@rayacorp.com wrote: Dear All , I am facing a problem to register br1 as MGCP on the CUCM The problem is when I use the T1 as H323 is working normally but when I need to use it as MGCP there is a problem in q921 As L2 be TEI Assigned forever . I tried no isdn bind-l3 ccm The output of the debug isdn q921 is below : *May 1 21:09:19.431: ISDN Se0/1/0:23 Q921: User RX - SABMEp sapi=0 tei=0 *May 1 21:09:19.431: ISDN Se0/1/0:23 Q921: S4_SABME: BACKHAULED vsc_wants_L2_up = FALSE *May 1 21:09:25.431: ISDN Se0/1/0:23 Q921: User RX - SABMEp sapi=0 tei=0 *May 1 21:09:25.431: ISDN Se0/1/0:23 Q921: S4_SABME: BACKHAULED vsc_wants_L2_up = FALSE *May 1 21:09:26.431: ISDN Se0/1/0:23 Q921: User RX - SABMEp sapi=0 tei=0 *May 1 21:09:26.431: ISDN Se0/1/0:23 Q921: S4_SABME: BACKHAULED vsc_wants_L2_up = FALSE *May 1 21:09:27.431: ISDN Se0/1/0:23 Q921: User RX - SABMEp sapi=0 tei=0 *May 1 21:09:27.431: ISDN Se0/1/0:23 Q921: S4_SABME: BACKHAULED vsc_wants_L2_up = FALSE *May 1 21:09:28.431: ISDN Se0/1/0:23 Q921: User RX - SABMEp sapi=0 tei=0 *May 1 21:09:28.431: ISDN Se0/1/0:23 Q921: S4_SABME: BACKHAULED vsc_wants_L2_up = FALSE *May 1 21:09:34.431: ISDN Se0/1/0:23 Q921: User RX - SABMEp sapi=0 tei=0 *May 1 21:09:34.431: ISDN Se0/1/0:23 Q921: S4_SABME: BACKHAULED vsc_wants_L2_up = FALSE *May 1 21:09:35.431: ISDN Se0/1/0:23 Q921: User RX - SABMEp sapi=0 tei=0 *May 1 21:09:35.431: ISDN Se0/1/0:23 Q921: S4_SABME: BACKHAULED vsc_wants_L2_up = FALSE *May 1 21:09:36.431: ISDN Se0/1/0:23 Q921: User RX - SABMEp sapi=0 tei=0 *May 1 21:09:36.431: ISDN Se0/1/0:23 Q921: S4_SABME: BACKHAULED vsc_wants_L2_up = FALSE *May 1 21:09:37.431: ISDN Se0/1/0:23 Q921: User RX - SABMEp sapi=0 tei=0 *May 1 21:09:37.431: ISDN Se0/1/0:23 Q921: S4_SABME: BACKHAULED vsc_wants_L2_up = FALSE Please find attached the PSTN ,BR1 configuration and image of the call manager for this GW. Can anyone help me to fulfill this problem? Thanks, Ahmed Ellboudy | CCNP, CCVP. Disclaimer: NOTICE The information contained in this message is confidential and is intended for the addressee(s) only. If you have received this message in error or there are any problems please notify the originator immediately. The unauthorized use, disclosure, copying or alteration of this message is strictly forbidden. Raya will not be liable for direct, special, indirect or consequential damages arising from alteration of the contents of this message by a third party or as a result of any malicious code or virus being passed on. Views expressed in this communication are not necessarily those of Raya.If you have received this message in error, please notify the sender immediately by email, facsimile or telephone and return and/or destroy the original message. br1mgcp.txt MGCP.jpg pstn.txt ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.PlatinumPlacement.com Disclaimer: NOTICE The information contained in this message is confidential and is intended for the addressee(s) only. If you have received this message in error or there are any problems please notify the originator immediately. The unauthorized use, disclosure, copying or alteration of this message is strictly forbidden. Raya will not be liable for direct, special, indirect or consequential damages arising from alteration of the contents of this message by a third party or as a result of any malicious code or virus being passed on. Views expressed in this communication are not necessarily those of Raya.If you have received this message in error, please notify the sender immediately by email, facsimile or telephone and return and/or destroy the original message.
Re: [OSL | CCIE_Voice] MGCP gateway - 1 ring disconnect (7965), fastbusy (7960)
Check your source interface (bind), CUCM must be registered with the same bind interface mentioned in the mgcp configuration. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Rrcrumm Sent: Friday, April 29, 2011 8:05 AM To: Adil Shaikh Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MGCP gateway - 1 ring disconnect (7965), fastbusy (7960) Hi Did you do a show isdn status? You may have to do a no isdn bind-l3 ccm manager on the serial interface Also no mgcp no mgcp bind control mgcp bind control source interface lo0 Mgcp Huh Rc Sent from my iPhone On Apr 28, 2011, at 8:19 PM, Adil Shaikh adil.sha...@gmail.commailto:adil.sha...@gmail.com wrote: hi all, i configured mgcp gateway on HQ RTR. when i call from PSTN to a 7965 phone, i get fastbusy after 1 ring. when i call from PSTN to a 7960 phone (which is set to auto answer after 2 rings), i get fastbusy straight away. so, i issued 'no mgcp' - waited for 60 sec to ensure it deregister and then issued 'mgcp' but the result does not change. i did this few times without success. so, i reloaded router and then everything worked fine. does anyone know the way to resolve this issue without reloading router? thanks -adil -- .. . . _7___|___|_|_|adil.sha...@gmail.commailto:adil.sha...@gmail.com . . ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.PlatinumPlacement.com * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ 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] RSVP based CAC
Have you set ip rsvp bandwidth in the serial interface? From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vinay Kumar6 Sent: Friday, April 29, 2011 9:58 AM To: Rogers Ochieng Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] RSVP based CAC Yes, it is there. Warm Regards, Vinay Kumar From: Rogers Ochieng rogersochi...@gmail.com To: Vinay Kumar6/India/IBM@IBMIN Cc: ccie_voice@onlinestudylist.com Date: 04/29/2011 10:25 AM Subject: Re: [OSL | CCIE_Voice] RSVP based CAC do you have this under your IOS dspfarm profile Mtp configuration rsvp codec pass-through On 29 April 2011 07:24, Vinay Kumar6 vinayjaisw...@in.ibm.commailto:vinayjaisw...@in.ibm.com wrote: Hi, Trying to configure Location based CAC using RSVP, have done the configuration buut it always says not enough bandwidth even though i have given ample bandwidth on the serial interfaces. Steps used to configure: Configured MTP on the HQ and Branches-Registered to HQ and Branch DP. Assigned the HQ and Branch MTPs to respective DPs using MRGL. Created Location for HQ and Branch and assigned to respective DPs. Reservation is mandatory. Codec used on between the regions is G729. Warm Regards, Vinay Kumar ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.platinumplacement.com/ * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ 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] RSVP based CAC
What is the codec between regions? Make sure you're using g729 and disable the g722 advertising and also the iLBC From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vinay Kumar6 Sent: Friday, April 29, 2011 11:28 AM Cc: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] RSVP based CAC Following is the config on the routers. interface Serial0/0/0.1 point-to-point ip address 10.10.1.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 401 ip rsvp bandwidth 40 dspfarm profile 2 mtp codec g729r8 codec pass-through rsvp maximum sessions software 4 associate application SCCP dspfarm profile 1 mtp codec g729r8 codec pass-through rsvp maximum sessions software 4 associate application SCCP interface Serial0/2/0.1 point-to-point ip address 10.10.1.2 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 501 ip rsvp bandwidth 40 Is there any step by step guide to configure or troubleshoot it? Warm Regards, Vinay Kumar From: Vinay Kumar6/India/IBM@IBMIN To: Rogers Ochieng rogersochi...@gmail.com Cc: ccie_voice@onlinestudylist.com Date: 04/29/2011 11:46 AM Subject: Re: [OSL | CCIE_Voice] RSVP based CAC Yes, it is there. Warm Regards, Vinay Kumar From: Rogers Ochieng rogersochi...@gmail.com To: Vinay Kumar6/India/IBM@IBMIN Cc: ccie_voice@onlinestudylist.com Date: 04/29/2011 10:25 AM Subject: Re: [OSL | CCIE_Voice] RSVP based CAC do you have this under your IOS dspfarm profile Mtp configuration rsvp codec pass-through On 29 April 2011 07:24, Vinay Kumar6 vinayjaisw...@in.ibm.commailto:vinayjaisw...@in.ibm.com wrote: Hi, Trying to configure Location based CAC using RSVP, have done the configuration buut it always says not enough bandwidth even though i have given ample bandwidth on the serial interfaces. Steps used to configure: Configured MTP on the HQ and Branches-Registered to HQ and Branch DP. Assigned the HQ and Branch MTPs to respective DPs using MRGL. Created Location for HQ and Branch and assigned to respective DPs. Reservation is mandatory. Codec used on between the regions is G729. Warm Regards, Vinay Kumar ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://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 * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ 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] MGCP gateway - 1 ring disconnect (7965), fastbusy (7960)
Have you checked the status of your MTP on CUCM (make sure It’s registred) Naoufal From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Naoufal Kerboute Sent: Friday, April 29, 2011 11:10 AM To: Rrcrumm; Adil Shaikh Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MGCP gateway - 1 ring disconnect (7965), fastbusy (7960) Check your source interface (bind), CUCM must be registered with the same bind interface mentioned in the mgcp configuration. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Rrcrumm Sent: Friday, April 29, 2011 8:05 AM To: Adil Shaikh Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MGCP gateway - 1 ring disconnect (7965), fastbusy (7960) Hi Did you do a show isdn status? You may have to do a no isdn bind-l3 ccm manager on the serial interface Also no mgcp no mgcp bind control mgcp bind control source interface lo0 Mgcp Huh Rc Sent from my iPhone On Apr 28, 2011, at 8:19 PM, Adil Shaikh adil.sha...@gmail.commailto:adil.sha...@gmail.com wrote: hi all, i configured mgcp gateway on HQ RTR. when i call from PSTN to a 7965 phone, i get fastbusy after 1 ring. when i call from PSTN to a 7960 phone (which is set to auto answer after 2 rings), i get fastbusy straight away. so, i issued 'no mgcp' - waited for 60 sec to ensure it deregister and then issued 'mgcp' but the result does not change. i did this few times without success. so, i reloaded router and then everything worked fine. does anyone know the way to resolve this issue without reloading router? thanks -adil -- .. . . _7___|___|_|_|adil.sha...@gmail.commailto:adil.sha...@gmail.com . . ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.PlatinumPlacement.com * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system
Re: [OSL | CCIE_Voice] RSVP based CAC
Have you checked the status of your MTP on CUCM (make sure It’s registred) Naoufal From: Vinay Kumar6 [mailto:vinayjaisw...@in.ibm.com] Sent: Friday, April 29, 2011 12:33 PM To: Naoufal Kerboute Cc: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] RSVP based CAC Codec is g729 and g722 is disabled in enterprise parameters. From the debug on the gateways I found a message which says RSVP Confirmation not required. Warm Regards, Vinay From: Naoufal Kerboute naou...@mhdinfotech.com To: Vinay Kumar6/India/IBM@IBMIN Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com, ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com Date: 04/29/2011 01:32 PM Subject: RE: [OSL | CCIE_Voice] RSVP based CAC What is the codec between regions? Make sure you’re using g729 and disable the g722 advertising and also the iLBC From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vinay Kumar6 Sent: Friday, April 29, 2011 11:28 AM Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] RSVP based CAC Following is the config on the routers. interface Serial0/0/0.1 point-to-point ip address 10.10.1.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 401 ip rsvp bandwidth 40 dspfarm profile 2 mtp codec g729r8 codec pass-through rsvp maximum sessions software 4 associate application SCCP dspfarm profile 1 mtp codec g729r8 codec pass-through rsvp maximum sessions software 4 associate application SCCP interface Serial0/2/0.1 point-to-point ip address 10.10.1.2 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 501 ip rsvp bandwidth 40 Is there any step by step guide to configure or troubleshoot it? Warm Regards, Vinay Kumar From: Vinay Kumar6/India/IBM@IBMIN To: Rogers Ochieng rogersochi...@gmail.com Cc: ccie_voice@onlinestudylist.com Date: 04/29/2011 11:46 AM Subject: Re: [OSL | CCIE_Voice] RSVP based CAC Yes, it is there. Warm Regards, Vinay Kumar From: Rogers Ochieng rogersochi...@gmail.com To: Vinay Kumar6/India/IBM@IBMIN Cc: ccie_voice@onlinestudylist.com Date: 04/29/2011 10:25 AM Subject: Re: [OSL | CCIE_Voice] RSVP based CAC do you have this under your IOS dspfarm profile Mtp configuration rsvp codec pass-through On 29 April 2011 07:24, Vinay Kumar6 vinayjaisw...@in.ibm.commailto:vinayjaisw...@in.ibm.com wrote: Hi, Trying to configure Location based CAC using RSVP, have done the configuration buut it always says not enough bandwidth even though i have given ample bandwidth on the serial interfaces. Steps used to configure: Configured MTP on the HQ and Branches-Registered to HQ and Branch DP. Assigned the HQ and Branch MTPs to respective DPs using MRGL. Created Location for HQ and Branch and assigned to respective DPs. Reservation is mandatory. Codec used on between the regions is G729. Warm Regards, Vinay Kumar ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://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 * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system
Re: [OSL | CCIE_Voice] RSVP based CAC
Can you paste the full config of your HQ and BR1 routers. If you remove the rsvp, you can make calls between HQ and BR1? From: Vinay Kumar6 [mailto:vinayjaisw...@in.ibm.com] Sent: Friday, April 29, 2011 3:10 PM To: Naoufal Kerboute Cc: ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] RSVP based CAC Yes, It is registered. Apr 29 10:47:45.600: RSVP 10.10.11.1_17198-10.10.10.1_17944[0.0.0.0]: RESV: no path information for 10.10.10.1 Where 10.10.10.1 is HQ MTP and 10.10.11.1 is BR MTP IP. because of which the RSVP is not getting initiated. Not sure how to fix this. Warm Regards, Vinay Kumar MTS-Remote Support Centre. IBM India Private Limited, Subramanya Arcade1, 12, Bannerghatta Main Road, Bangalore 560 029 (India) Telephone: Direct +91-80-40683977, Board +91-80-4068 3000, Extn: 83977, Fax +91-80-26787711 Email : vinayjaisw...@in.ibm.commailto:vinayjaisw...@in.ibm.com From: Naoufal Kerboute naou...@mhdinfotech.com To: Vinay Kumar6/India/IBM@IBMIN Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com, ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com Date: 04/29/2011 04:21 PM Subject: RE: [OSL | CCIE_Voice] RSVP based CAC Have you checked the status of your MTP on CUCM (make sure It’s registred) Naoufal From: Vinay Kumar6 [mailto:vinayjaisw...@in.ibm.com] Sent: Friday, April 29, 2011 12:33 PM To: Naoufal Kerboute Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] RSVP based CAC Codec is g729 and g722 is disabled in enterprise parameters. From the debug on the gateways I found a message which says RSVP Confirmation not required. Warm Regards, Vinay From: Naoufal Kerboute naou...@mhdinfotech.com To: Vinay Kumar6/India/IBM@IBMIN Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com, ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com Date: 04/29/2011 01:32 PM Subject: RE: [OSL | CCIE_Voice] RSVP based CAC What is the codec between regions? Make sure you’re using g729 and disable the g722 advertising and also the iLBC From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vinay Kumar6 Sent: Friday, April 29, 2011 11:28 AM Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com; ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] RSVP based CAC Following is the config on the routers. interface Serial0/0/0.1 point-to-point ip address 10.10.1.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 401 ip rsvp bandwidth 40 dspfarm profile 2 mtp codec g729r8 codec pass-through rsvp maximum sessions software 4 associate application SCCP dspfarm profile 1 mtp codec g729r8 codec pass-through rsvp maximum sessions software 4 associate application SCCP interface Serial0/2/0.1 point-to-point ip address 10.10.1.2 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 501 ip rsvp bandwidth 40 Is there any step by step guide to configure or troubleshoot it? Warm Regards, Vinay Kumar From: Vinay Kumar6/India/IBM@IBMIN To: Rogers Ochieng rogersochi...@gmail.com Cc: ccie_voice@onlinestudylist.com Date: 04/29/2011 11:46 AM Subject: Re: [OSL | CCIE_Voice] RSVP based CAC Yes, it is there. Warm Regards, Vinay Kumar From: Rogers Ochieng rogersochi...@gmail.com To: Vinay Kumar6/India/IBM@IBMIN Cc: ccie_voice@onlinestudylist.com Date: 04/29/2011 10:25 AM Subject: Re: [OSL | CCIE_Voice] RSVP based CAC do you have this under your IOS dspfarm profile Mtp configuration rsvp codec pass-through On 29 April 2011 07:24, Vinay Kumar6 vinayjaisw...@in.ibm.commailto:vinayjaisw...@in.ibm.com wrote: Hi, Trying to configure Location based CAC using RSVP, have done the configuration buut it always says not enough bandwidth even though i have given ample bandwidth on the serial interfaces. Steps used to configure: Configured MTP on the HQ and Branches-Registered to HQ and Branch DP. Assigned the HQ and Branch MTPs to respective DPs using MRGL. Created Location for HQ and Branch and assigned to respective DPs. Reservation is mandatory. Codec used on between the regions is G729. Warm Regards, Vinay Kumar ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.platinumplacement.com
[OSL | CCIE_Voice] Proctor is DOWN
Guys, Proctor is DOWN? Naoufal. * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ 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] CUPS integration with LDAP
Just go ahead, no need for LDAP From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Nizar Abuseni Sent: Wednesday, April 27, 2011 10:25 AM To: ccie_voice Subject: [OSL | CCIE_Voice] CUPS integration with LDAP Hello, I have question regarding integration CUPS server with LDAP (AD), is it necessary to have the CUCM also integrated with LDAP (AD)? Also can I use CUCM as LDAP? if yes will the directory search work? and can I add users? Am trying to find a way so that I don’t want to integrate CUCM with AD, because I want to use different user ID’s in CUCM than user ID’s in AD. So if I don’t use the AD at all will be better. Best regards, Nizar * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ 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] workbook 2 lab 8
Hi, For unity connection check Redirection Header Delivery at SIP trunk configuration (outbound). For CUE and SRST, please post your config. Rgds, Naoufal From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cristobal Priego Sent: Saturday, April 23, 2011 7:56 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] workbook 2 lab 8 Hello all I just finished my session and i was doing lab 8 from workbooks 2 something that i couldn't get to work properly was the unity integration through sip trunk the ring no answer and the busy were playing enter your id followed by pound instead of sorry extension ... is not available record your message at the tone i followed the proctor-guide and still it' didn't work also i couldn't get the MWI to work on CUE sip phones only my config looked like this voice register dn 1 number 3002 call-forward b2bua busy 3600 call-forward b2bua noan 3600 timeout 12 mwi name br2 phone 4 sip-ua mwi-server ipv4:10.10.202.2 i had the unsolicited notify enabled on the cue gui when i was doing a refresh of the mwi i saw unity express trying to ring my extensions on the default mwi extensions so i went ahead and configured the ephone dn's for mwi still didn't work also my sip srst didn't work i kept getting this error Apr 23 07:46:38.660: //-1//SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.202.1:5060;branch=z9hG4bKe43aa9be From: sip:1002@10.10.201.1mailto:sip%3A1002@10.10.201.1;tag=001ae22b12c500077cb0194d-395f52ee To: sip:1002@10.10.201.1mailto:sip%3A1002@10.10.201.1;tag=17DB834-A33 Date: Sat, 23 Apr 2011 07:46:38 GMT Call-ID: 001ae22b-12c50006-85a48968-9fe91895@192.168.11.12mailto:001ae22b-12c50006-85a48968-9fe91895@192.168.11.12 Server: Cisco-SIPGateway/IOS-12.x CSeq: 105 REGISTER Content-Length: 0 my sip srst looked like this voice register pool 1 id network 10.10.201.0 mask 255.255.255.0 cor incoming ld-css default call-forward b2bua busy 5600 call-foward b2bua noan 5600 timeout 13 codec g711u voice register global max-pool 2 max-dn 2 please help me out, thank you * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ 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] AAR question
Hi, First try to make a call from HQ and BR1 phones to CUE directly without AAR, once is OK then assign the AAR group and Css to all phones and CTI ports, set the mask on the voicemail profile and test. Make sure to have RP that can route the external phone number musk trough the local gateway. I think it should work. Rgds, Naoufal From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cristobal Priego Sent: Monday, April 18, 2011 7:51 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] AAR question Hello All, I have a question i was doing the workbook 2 lab 7 and i had a questions that involved AAR and i got stuck br1, br2, hq were all ucm sites when the bandwidth between br2 and the other 2 sites (Br1, hq) wasn't enough to make the call AAR should kick in the br2 phones had a mailbox with CUE (cue was registered to ucm) I enabled AAR i setup AAR, the AAR group applied everywhere the AAR css, applied everywhere as well i had the External Phone Number mask set i created my route pattern being very specific and did all the digit manipulation so the call will succeed when i went to locations and reduced the BW to 20 and resynched the BW when i was calling br2 phone from br1 or hq. i was getting the AA for CUE and the called number on the pri was the external phone number mask assigned to the CTI Route Point and on the display on br1/hq phone was saying not enough bw, rerouting however when i went ahead and removed the call forward settings on BR2 phone and when I was trying to call br2 phone by dialing the internal extension ... i was getting busy no messages on the phones could you please help me understand why ? thank you * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] TCS Capability Exchange
Thank you guys, I found it H245SessionEstablishedFailure | waitForCapabilitiesExchange In the beginning I ended the call once I discovered the issue, so I have to wait for the call disconnect to get this error in the debug. It was my mistake. Thanks again Naoufal -Original Message- From: Farkas Péter [mailto:wormh...@sch.bme.hu] Sent: Monday, April 18, 2011 12:14 PM To: George Goglidze Cc: Naoufal Kerboute; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] TCS Capability Exchange It is rather a timer expiration since either endpoint will send TCS. Peter - Original Message - From: George Goglidze gogli...@gmail.com Date: Sunday, April 17, 2011 6:35 pm Subject: Re: [OSL | CCIE_Voice] TCS Capability Exchange To: Naoufal Kerboute naou...@mhdinfotech.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com I don't have a call manager available right now, but if you keep searching for h245 it said something about capability failure if I remember correctly. Attach the trace file if you have it, I'll have a look. Sent from my iPad On 17 Apr 2011, at 05:54, Naoufal Kerboute naou...@mhdinfotech.commailto:naou...@mhdinfotech.com wrote: Hi guys, I'm working on CUBE, and I'm facing the TCS issue, I know that I have to uncheck wait for Far End H.245 Terminal Capability Set, but I'm looking how to identify this in the SDL traces. Anyone know the exact word who describe the issue in the logs? Thanks a lot Naoufal ** ** ** *** * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * ** ** ** *** ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] AAR question
Don't forget to add external phone number mask to CTI ports and also transcoder in your BR2 for CUE From: Cristobal Priego [mailto:cristobalpri...@gmail.com] Sent: Tuesday, April 19, 2011 7:56 AM To: Naoufal Kerboute Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] AAR question thank you I'm going to try it this Friday 2011/4/17 Naoufal Kerboute naou...@mhdinfotech.commailto:naou...@mhdinfotech.com Hi, First try to make a call from HQ and BR1 phones to CUE directly without AAR, once is OK then assign the AAR group and Css to all phones and CTI ports, set the mask on the voicemail profile and test. Make sure to have RP that can route the external phone number musk trough the local gateway. I think it should work. Rgds, Naoufal From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cristobal Priego Sent: Monday, April 18, 2011 7:51 AM To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] AAR question Hello All, I have a question i was doing the workbook 2 lab 7 and i had a questions that involved AAR and i got stuck br1, br2, hq were all ucm sites when the bandwidth between br2 and the other 2 sites (Br1, hq) wasn't enough to make the call AAR should kick in the br2 phones had a mailbox with CUE (cue was registered to ucm) I enabled AAR i setup AAR, the AAR group applied everywhere the AAR css, applied everywhere as well i had the External Phone Number mask set i created my route pattern being very specific and did all the digit manipulation so the call will succeed when i went to locations and reduced the BW to 20 and resynched the BW when i was calling br2 phone from br1 or hq. i was getting the AA for CUE and the called number on the pri was the external phone number mask assigned to the CTI Route Point and on the display on br1/hq phone was saying not enough bw, rerouting however when i went ahead and removed the call forward settings on BR2 phone and when I was trying to call br2 phone by dialing the internal extension ... i was getting busy no messages on the phones could you please help me understand why ? thank you * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ 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] TCS Capability Exchange
Hi guys, I'm working on CUBE, and I'm facing the TCS issue, I know that I have to uncheck wait for Far End H.245 Terminal Capability Set, but I'm looking how to identify this in the SDL traces. Anyone know the exact word who describe the issue in the logs? Thanks a lot Naoufal * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] QOS Class Based Shaping
Hi guys, I'm working on QoS Vol2 lab7, when I assigned the class to serial interface I got the below message: HQ-RTR(config)#interface Serial0/0/1:0.1 point-to-point HQ-RTR(config-subif)#bandw HQ-RTR(config-subif)#bandwidth 384 HQ-RTR(config-subif)# frame-relay interface-dlci 201 HQ-RTR(config-fr-dlci)#cla HQ-RTR(config-fr-dlci)#class BR1 I/f shape class SIG requested bandwidth 18 (kbps), available only 12 (kbps) I/f shape class SIG requested bandwidth 18 (kbps), available only 12 (kbps) Below my QoS Config class-map match-any RTP match protocol rtp audio match protocol rtcp class-map match-any SIG match protocol skinny match protocol h323 match protocol mgcp match protocol sip match protocol rsvp ! ! policy-map WAN-EDGE class RTP compress header ip rtp priority 24 class SIG bandwidth 18 policy-map Shape-BR1 class class-default shape average 36480 3648 0 service-policy WAN-EDGE ! map-class frame-relay BR1 frame-relay fragment 480 service-policy output Shape-BR1 Any Ideas? Thanks NAOUFAL * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Voice mail in CUPS
Not the CUCM password you have tu use the unity connection user web password From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Erwan Erwan Sent: Sunday, April 17, 2011 8:10 AM To: Roger Carpio; Vik Malhi Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Voice mail in CUPS hmm yes i tried this too: br1ph1 , password :cisco , from UCM End User --- On Sun, 4/17/11, Vik Malhi vma...@ipexpert.commailto:vma...@ipexpert.com wrote: From: Vik Malhi vma...@ipexpert.commailto:vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] Voice mail in CUPS To: Roger Carpio roger.car...@gmail.commailto:roger.car...@gmail.com Cc: Erwan Erwan e_er...@yahoo.commailto:e_er...@yahoo.com, ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Received: Sunday, April 17, 2011, 11:39 AM Also remember it is not the voicemail password but rather the web application password that the CUPC will be using. -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.comhttp://ca.mc1205.mail.yahoo.com/mc/compose?to=vma...@ipexpert.com Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chathttp://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/communitieshttp://www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.comhttp://www.ipexpert.com/ http://www.ipexpert.com/ On Apr 16, 2011, at 18:53, Roger Carpio roger.car...@gmail.comhttp://ca.mc1205.mail.yahoo.com/mc/compose?to=roger.car...@gmail.com wrote: Before adding the user to UC; did you configure the Allow Users to Access Voice Mail Using an IMAP Client option in COS? On Sat, Apr 16, 2011 at 6:32 PM, Erwan Erwan e_er...@yahoo.comhttp://ca.mc1205.mail.yahoo.com/mc/compose?to=e_er...@yahoo.com wrote: hi all, can someone advice, what i miss in CUPS voicemail. I kept geeting this error from Show Server Health in CUPC client Failed to Connect - Invalid Credentials or Account Locked I verified user name and password for voicemail is working in Phone itself tks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] B-ACD Not working
Hi, Add service aa flash:app-b-acd-aa-3.0.0.2.tcl param number-of-hunt-grps 3 and correct the number of hunt group in ACD service (I guess you have 3 not 4) Hope this will help Naoufal From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of mgscip Sent: Wednesday, April 13, 2011 5:36 PM To: ccie Subject: [OSL | CCIE_Voice] B-ACD Not working Hi , We tested with B-ACD in CME . whenever we dial the pilot number call disconnect. Config application service aa flash:app-b-acd-aa-3.0.0.2.tcl paramspace english index 1 param menu-timeout 1 param dial-by-extension-option 1 param handoff-string aa paramspace english language en param max-time-vm-retry 2 param aa-pilot 7500 paramspace english location flash: param second-greeting-time 1 param welcome-prompt _bacd_welcome.au paramspace english prefix en param service-name ACD service ACD flash:app-b-acd-3.0.0.2.tcl paramspace english language en paramspace english index 0 param aa-hunt1 7001 param aa-hunt2 7002 param number-of-hunt-grps 4 param aa-hunt3 7003 Dial-peer dial-peer voice 7500 voip service aa destination-pattern 7500 session target ipv4:192.168.1.100 incoming called-number 7500 dtmf-relay h245-alphanumeric codec g711ulaw I have verified that all the audio files uploade in the flash. MoH working for IP Phones. When i check show call application session , it tries to establish session but it end session immediately * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] UCCX scripting (Unity connection for Holiday query)
Hi guys, I'm working on a UCCX script (Vol2 Lab7) that permit to check the holiday in a xml file and then decides either to terminate or accept the call. I'm searching if there is another way to use Unity Connection for the holiday query instead of docs. Any ideas? Thanks a lot Naoufal * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 4.2)
Hi, You have to register the br2 with the UCME zone not the VIA zone. Remove h323-gateway voip id VIA ipaddr 172.1.254.1 1719 and replace it with h323-gateway voip id UCME ipaddr 172.1.254.1 1719 Thanks Naoufal -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Alex Goh Sent: Saturday, April 09, 2011 9:43 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 4.2) Hi Guys, I'm trying to get the solutions for question 4.2 to work, but apparently I'm missing something and hope someone can help. I've search thru the list but doesn't really found a solution work for my case. The issue I've encounter are when HQ phone 5001 calling BR2 phone 3003, 3003 ring, but when i tried to answered, the call drop. I know it might be related to codec issue, but I've my HQ-RTR configured with Xcoder which it is up and active but the call still failing. I also did have the trunk in cucm Wait for Far End H.245 Terminal Capability Set unchecked. once things I notice is that, my call doesn't seems get re-originated on the cube router to BR2 router, what I see during ringing state my show gatekeeper endpoint show the call is directly from the CUCM to BR2 It is only 2 call legs instead of 4 (see below). hm, what have I missed? Some Info: HQ Router (R1) interface Loopback0 ip address 172.1.254.1 255.255.255.255 h323-gateway voip interface h323-gateway voip id VIA ipaddr 172.1.254.1 1719 h323-gateway voip h323-id R1 h323-gateway voip bind srcaddr 172.1.254.1 gatekeeper zone local UCM 172.1.254.1 zone local UCME outvia VIA zone local VIA zone prefix UCME 3... gw-type-prefix 1#* default-technology no shutdown dial-peer voice 30 voip destination-pattern 3... session target ras codec g711ulaw ! dial-peer voice 31 voip incoming called-number 3... Total number of active calls = 1. GATEKEEPER CALL INFO LocalCallIDAge(secs) BW 511-32797 6 16(Kbps) Endpt(s): Alias E.164Addr src EP: gk_trunk_25001 CallSignalAddr Port RASSignalAddr Port 172.1.10.20 38233 172.1.10.20 32795 Endpt(s): Alias E.164Addr dst EP: R33003 CallSignalAddr Port RASSignalAddr Port 172.3.254.1 1720 172.3.254.1 49395 GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags --- - --- - - - 172.1.10.10 47142 172.1.10.10 32838 UCM VOIP-GW H323-ID: gk_trunk_1 Voice Capacity Max.= Avail.= Current.= 0 172.1.10.20 38233 172.1.10.20 32795 UCM VOIP-GW H323-ID: gk_trunk_2 Voice Capacity Max.= Avail.= Current.= 0 172.1.254.1 1720 172.1.254.2 56974 VIA H323-GW H323-ID: R1 Voice Capacity Max.= Avail.= Current.= 0 172.3.254.1 1720 172.3.254.1 49395 VIA H323-GW H323-ID: R3 Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 4 R1(config-if)#do sh gatek gw GATEWAY TYPE PREFIX TABLE = Prefix: 1#*(Default gateway-technology) Zone UCM master gateway list: 172.1.10.20:38233 gk_trunk_2 172.1.10.10:47142 gk_trunk_1 Zone VIA master gateway list: 172.3.254.1:1720 R3 172.1.254.2:1720 R1 BR2 Router (R2) interface Loopback0 ip address 172.3.254.1 255.255.255.255 h323-gateway voip interface h323-gateway voip id VIA ipaddr 172.1.254.1 1719 h323-gateway voip h323-id R3 h323-gateway voip tech-prefix 1# h323-gateway voip bind srcaddr 172.3.254.1 dial-peer voice 10 voip incoming called-number 3... dtmf-relay rtp-nte codec g711ulaw ! CUCM Trunk the trunk was assign a separate DP with a region that using G729 when calling HQ and BR2. Regards, Alex ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part
Re: [OSL | CCIE_Voice] UCCX Script (Redirect call based on the calling number)
The + symbol is a string so it can be match. My script is working if I set the condition like If Calling Number == “+3434141891” then redirect call to 5001 But I’m looking for a way to reroute all calls coming from area +34 to 5001 Naoufal From: bkvalent...@gmail.com [mailto:bkvalent...@gmail.com] Sent: Sunday, April 10, 2011 4:40 PM To: Naoufal Kerboute; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] UCCX Script (Redirect call based on the calling number) I believe uccx does not understand the + symbol. - Reply message - From: Naoufal Kerboute naou...@mhdinfotech.com Date: Sun, Apr 10, 2011 7:21 am Subject: [OSL | CCIE_Voice] UCCX Script (Redirect call based on the calling number) To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Dear gents, I'm working on UCCX section and I'm trying to reroute some calls coming from Spain (+34) to a specific extension. I've setup the script and it's working only if I set the calling number variable to full Spain PSTN number, but let take the case for many number from Spain. How can I reroute calls coming from spain to a specific extension (I don't want to much the full muber, I want to much only calling number start with +34) Any ideas? Thanks a lot Naoufal * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 4.2)
Hi, For supplementary you have to setup an MTP, For the call drop try to enable Inbound Fast Start Naoufal -Original Message- From: Alex Goh [mailto:ncsalex@gmail.com] Sent: Sunday, April 10, 2011 6:29 PM To: Naoufal Kerboute Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 4.2) Hi All, Thanks very much for the reply. The issue is due to my mistake that registering BR2 to wrong zone. Now the CUCM Call to BR2 is working fine except the supplementary service e.g hold, Moh doesn't work, do I need MTP for this? also, calling from BR2 Sip phone to CUCM is failling, phone ring, but when answered, it dropped. my Sip phone is using G729 codec, do I still need MTP on BR2 in this case? Thanks Regards, Alex On Sun, Apr 10, 2011 at 2:19 AM, Naoufal Kerboute naou...@mhdinfotech.com wrote: Hi, You have to register the br2 with the UCME zone not the VIA zone. Remove h323-gateway voip id VIA ipaddr 172.1.254.1 1719 and replace it with h323-gateway voip id UCME ipaddr 172.1.254.1 1719 Thanks Naoufal -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Alex Goh Sent: Saturday, April 09, 2011 9:43 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 4.2) Hi Guys, I'm trying to get the solutions for question 4.2 to work, but apparently I'm missing something and hope someone can help. I've search thru the list but doesn't really found a solution work for my case. The issue I've encounter are when HQ phone 5001 calling BR2 phone 3003, 3003 ring, but when i tried to answered, the call drop. I know it might be related to codec issue, but I've my HQ-RTR configured with Xcoder which it is up and active but the call still failing. I also did have the trunk in cucm Wait for Far End H.245 Terminal Capability Set unchecked. once things I notice is that, my call doesn't seems get re-originated on the cube router to BR2 router, what I see during ringing state my show gatekeeper endpoint show the call is directly from the CUCM to BR2 It is only 2 call legs instead of 4 (see below). hm, what have I missed? Some Info: HQ Router (R1) interface Loopback0 ip address 172.1.254.1 255.255.255.255 h323-gateway voip interface h323-gateway voip id VIA ipaddr 172.1.254.1 1719 h323-gateway voip h323-id R1 h323-gateway voip bind srcaddr 172.1.254.1 gatekeeper zone local UCM 172.1.254.1 zone local UCME outvia VIA zone local VIA zone prefix UCME 3... gw-type-prefix 1#* default-technology no shutdown dial-peer voice 30 voip destination-pattern 3... session target ras codec g711ulaw ! dial-peer voice 31 voip incoming called-number 3... Total number of active calls = 1. GATEKEEPER CALL INFO LocalCallID Age(secs) BW 511-32797 6 16(Kbps) Endpt(s): Alias E.164Addr src EP: gk_trunk_2 5001 CallSignalAddr Port RASSignalAddr Port 172.1.10.20 38233 172.1.10.20 32795 Endpt(s): Alias E.164Addr dst EP: R3 3003 CallSignalAddr Port RASSignalAddr Port 172.3.254.1 1720 172.3.254.1 49395 GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags --- - --- - - - 172.1.10.10 47142 172.1.10.10 32838 UCM VOIP-GW H323-ID: gk_trunk_1 Voice Capacity Max.= Avail.= Current.= 0 172.1.10.20 38233 172.1.10.20 32795 UCM VOIP-GW H323-ID: gk_trunk_2 Voice Capacity Max.= Avail.= Current.= 0 172.1.254.1 1720 172.1.254.2 56974 VIA H323-GW H323-ID: R1 Voice Capacity Max.= Avail.= Current.= 0 172.3.254.1 1720 172.3.254.1 49395 VIA H323-GW H323-ID: R3 Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 4 R1(config-if)#do sh gatek gw GATEWAY TYPE PREFIX TABLE = Prefix: 1#* (Default gateway-technology) Zone UCM master gateway list: 172.1.10.20:38233 gk_trunk_2 172.1.10.10:47142 gk_trunk_1 Zone VIA master gateway list: 172.3.254.1:1720 R3 172.1.254.2:1720 R1 BR2 Router (R2) interface Loopback0 ip address 172.3.254.1 255.255.255.255 h323-gateway voip interface h323-gateway voip id VIA ipaddr 172.1.254.1 1719 h323-gateway voip h323-id R3 h323-gateway voip tech-prefix 1# h323-gateway voip bind srcaddr 172.3.254.1 dial-peer voice 10 voip incoming called-number 3... dtmf-relay rtp-nte codec g711ulaw ! CUCM Trunk the trunk
Re: [OSL | CCIE_Voice] UCCX Script (Redirect call based on the calling number)
Thanks Roger. I was looking for the function StartsWith(+34). You are a great man :D From: Rogers Ochieng [mailto:rogersochi...@gmail.com] Sent: Sunday, April 10, 2011 7:57 PM To: Naoufal Kerboute Cc: bkvalent...@gmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] UCCX Script (Redirect call based on the calling number) Looks similar to requirement of IPExpert Workbook 1 Lab 12A - 12.2 On 10 April 2011 15:40, Naoufal Kerboute naou...@mhdinfotech.commailto:naou...@mhdinfotech.com wrote: The + symbol is a string so it can be match. My script is working if I set the condition like If Calling Number == +3434141891 then redirect call to 5001 But I'm looking for a way to reroute all calls coming from area +34 to 5001 Naoufal From: bkvalent...@gmail.commailto:bkvalent...@gmail.com [mailto:bkvalent...@gmail.commailto:bkvalent...@gmail.com] Sent: Sunday, April 10, 2011 4:40 PM To: Naoufal Kerboute; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] UCCX Script (Redirect call based on the calling number) I believe uccx does not understand the + symbol. - Reply message - From: Naoufal Kerboute naou...@mhdinfotech.commailto:naou...@mhdinfotech.com Date: Sun, Apr 10, 2011 7:21 am Subject: [OSL | CCIE_Voice] UCCX Script (Redirect call based on the calling number) To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Dear gents, I'm working on UCCX section and I'm trying to reroute some calls coming from Spain (+34) to a specific extension. I've setup the script and it's working only if I set the calling number variable to full Spain PSTN number, but let take the case for many number from Spain. How can I reroute calls coming from spain to a specific extension (I don't want to much the full muber, I want to much only calling number start with +34) Any ideas? Thanks a lot Naoufal * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have
Re: [OSL | CCIE_Voice] SRST mode with HSRP
Hi, Add the secondary address (which is your backup gw) in the telephony-services. Rgds Naoufal From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Julien Krieger Sent: Thursday, April 07, 2011 8:17 PM To: Glen Cobby; George Goglidze Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SRST mode with HSRP Hi guys, Thank you for your answers. George, I am more looking for a backup solution at this tima than load balancing. For load balancing, I could still use 2 HRSP instances and use 2 DP with 2 HSRP vip. With it, i will have both solution. Julien 2011/4/7 Glen Cobby g...@positivenetworks.co.ukmailto:g...@positivenetworks.co.uk I do that on sites with two gateways and it works fine. Thanks Glen On 7 Apr 2011, at 14:57, Julien Krieger krieger.jul...@gmail.commailto:krieger.jul...@gmail.com wrote: Hi guys, I am running a few tests to clear thing up on SRST with HSRP. I have 1 remote site with 2 local gateways. Theses gateways must act as SRST should my wan goes down. I would usually create 2 Device Pools with 1 of the 2 gateways as its SRST reference. But what if I gateway go down? What I would like to see is if I could use an HSRP ip add as my SRST reference into my 1 device pool. Do you think it would work? Julien ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Debug Gatekeeper trunk call - codec mismatch
You can see the bandwidth requested in the debug From: Rogers Ochieng [mailto:rogersochi...@gmail.com] Sent: Tuesday, April 05, 2011 10:02 AM To: Naoufal Kerboute Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Debug Gatekeeper trunk call - codec mismatch I ran debug gatekeeper call 10 but did know how to pick out the codec issue which i created by having PSTN phone on strict G711 while my CUCM trunk to HQ GK is on G729. Call rings then drop when i pickup, if i change the trunk or CUCM h225 trunk to use same codec it works. I need to pickout the error or the error code. I'll try out voice ccapi inout On 5 April 2011 07:13, Naoufal Kerboute naou...@mhdinfotech.commailto:naou...@mhdinfotech.com wrote: You can run: debug gatekeeper call 10 or debug voice ccapi inout NaoufaL From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Rogers Ochieng Sent: Tuesday, April 05, 2011 6:55 AM To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Debug Gatekeeper trunk call - codec mismatch Which debug output will show any codec mismatch? I know that i need to check for codec as one problem if a call between two endpoints get drop on answer. I need an expert level debug i can send to an ITSP and tell them, here's the mismatch. * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] lab 5 every location with SIP TRUNK
I like you Jonny :) From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jonny Mendas Sent: Monday, April 04, 2011 11:00 AM To: v.c...@yahoo.com; a...@ipcomconsult.com; jbr...@tsginc.biz; ccie_voice-boun...@onlinestudylist.com; gogli...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] lab 5 every location with SIP TRUNK Hi Guys, I can see lot of guys are freshers out here! Previously people pass the lab by making back to back attempts and when they attempt they remember the labs and get it then they make the solutions and attempt when there is no ipexpert,ine or 360 so lets say 3 attempts they fail they get all the lab and pass on 4th attempt less CCIE but more investment... The same things real labs are giving u now... So as smart work save yr 3 times attempt money and get it and make yrself pass Even there are guys who cannot pass from real labs but CCIE is respectful and you need hard core knowledge for the same even on real labs. But if you dont have that!!! then make 5 attempts get the lab and then pass but again u are doing the same thing ah ha... but with a stupid way!!! I am sure now you guys got the answer To reach till CCIE its default you require core knowledge there is no one in the world who just close yr eyes and attempt and make it pass atleast not in VOICE So now please do not speak like junk or freshers and update if you will get lab 5 :) Thankyou Date: Sun, 3 Apr 2011 20:57:07 -0700 From: v.c...@yahoo.commailto:v.c...@yahoo.com To: a...@ipcomconsult.commailto:a...@ipcomconsult.com; jbr...@tsginc.bizmailto:jbr...@tsginc.biz; ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com; gogli...@gmail.commailto:gogli...@gmail.com CC: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] lab 5 every location with SIP TRUNK I agree with Alex. In the exam they are doing a lot of bad things to prevent you from passing the exam not to measure the your skills. Regarding George message, I respect your message and I am not happy to hear what I said in my previous message about passing CCIE using real lab, yes I agree with you it is very bad to pass CCIE depending on real lab only. but for me I am considering real lab like other scenarios ( I will buy and study it like when I am studying IP Expert WB) for sure I will not just study the real lab and go to the exam, no I will prepare for the exam, I am going through the blue print topic by topic, I am reading to much in Cisco Press books. I have the complete solution of IP Expert I spent a lot of hours on racks and my home lab. but at the end I am going to take a look on real lab. I need to pass. I can not see my company respect some guys more than me just because they are CCIE and I have more experience and knowledge. Regards, From: Alex a...@ipcomconsult.com To: Justin Brady jbr...@tsginc.biz; ccie_voice-boun...@onlinestudylist.com; George Goglidze gogli...@gmail.com; Ccie Voice v.c...@yahoo.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Mon, April 4, 2011 6:32:54 AM Subject: Re: [OSL | CCIE_Voice] lab 5 every location with SIP TRUNK Guys, Most of the people on this list are trying to make way to their numbers without cheating. I also failed a few times but never resorted to real labs. However sometimes I I feel that cisco is cheating on me. Do u think R and S troubleshooting should be a part of Voice exam? Do u think that forcing u to use workaronds instead of normal tools available in real live adds any value to the certification? Don't u think that if cisco wants to test our troubleshoting skills they have to allocate some points and time for it but not brake something which u can possibly find only at the end of the exam when testing everything? I didn't have any problems configuring anything they asked me to. I also managed to resolve all the tricks they prepared for me (except of one for which I spent hours to research and recreate but still dono how did they brake it). However these hidden things kill ur time and as the result u don't have enough to verify everything and lose points because of typo etc. So, don't blame the guy much, he is just trying to cheat on cheater. Sent from my BlackBerry Wireless Handheld -Original Message- From: Justin Brady jbr...@tsginc.bizmailto:jbr...@tsginc.biz Sender: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com Date: Mon, 4 Apr 2011 00:42:41 To: George Goglidzegogli...@gmail.commailto:gogli...@gmail.com; Ccie Voicev.c...@yahoo.commailto:v.c...@yahoo.com Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.comccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] lab 5 every location with SIP TRUNK
Re: [OSL | CCIE_Voice] RSVP number of calls
For RSVP always consider the worst casr scenario for the first call, meaning if you're using g729 you have to reserve 40k for the first call and for g711 reserve 96k Naoufal From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of CCIE for Me Sent: Tuesday, April 05, 2011 6:33 AM To: voice boy; OSL Questions Subject: Re: [OSL | CCIE_Voice] RSVP number of calls with g729 and rsvp always use 40k for the first call (worst case) + 24 each subsequent call. You don't need to create an HQ location since it is already in hubnone (well, you can but you are making more work for yourself). All other devices that have those locations will need to be account for too in your calculations. From: voice boymailto:voice...@hotmail.com Sent: Monday, April 04, 2011 9:57 PM To: OSL Questionsmailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] RSVP number of calls Hi, I have 2 doubts here ... When I set RSVP between 2-sites to permit 2-G729 calls It work with me first when i use 80 for rsvp bandwidth but trying 40k for the first and from the second call 24k ,, it also work with me So I built my calculations on this so 2-calls = 64k ,, 3-calls = 88k ,, 4-calls = 112k and so one or use 40k for first and second and each needed call ? Also I'll create two locations 2-sites [HQ and another site] Do i need to create 3rd location with no reservation with all other locations and assign this location to GK,MMOH,CUC-ports,UCCX-ports RPs that all these exist in HQ and may be affected if put in HQ location as phones ?? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Debug Gatekeeper trunk call - codec mismatch
You can run: debug gatekeeper call 10 or debug voice ccapi inout NaoufaL From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Rogers Ochieng Sent: Tuesday, April 05, 2011 6:55 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Debug Gatekeeper trunk call - codec mismatch Which debug output will show any codec mismatch? I know that i need to check for codec as one problem if a call between two endpoints get drop on answer. I need an expert level debug i can send to an ITSP and tell them, here's the mismatch. * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] MGCP Gateway registration Issue
Hi gents, Please I need your help for the below. I'm always facing problem to get the MGCP gateway up, always the first time register with the serial interface instead of the bind interface (loopback or voice vlan int) that I specified in the config. I've tried to remove the isdn bind-l3 ccm-manager and put it again, no mgcp and mgcp but it doesn't help until I remove sometime the hole config and do it again and again, reset... I'm missing something?? Thanks a lot. Naoufal * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 5 released
Are you serious? Why you mail is ccievoicelab5? From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccievoice Sent: Tuesday, March 22, 2011 4:38 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Lab 5 released Guys, Lab 5 released guys i have no words to say!! It was my 9 attempt and again i got fucX Just to inform there was lot of things which i got it never seen in life :) SIP Trunk , AAR , MVA , CAC all changed IPCC page was one full page, SIP trunk troubleshooting was full one page I got so depressed that i left the lab like that ): ): ): ): ): ): ): ): [http://sigads.rediff.com/RealMedia/ads/adstream_nx.ads/www.rediffmail.com/signatureline.htm@Middle]http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm@Middle? * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCIE PASSE
___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ -- Naoufal Kerboute Networks Service Manager Ki Wi kiwi.vo...@gmail.com, Bill Lake whl...@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] FRF.12 - Recommendation: Set the CIR to 95 percent of the PVC contracted speed.
I guess you can keep 960 From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Friderich Claude Sent: Thursday, March 17, 2011 8:06 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] FRF.12 - Recommendation: Set the CIR to 95 percent of the PVC contracted speed. Hello Guys, If they ask us to configure 95 percent of the PVC with a bandwidth of 768k for example (FRF12) What value should I put for the fragmentation size meaning best practice is 10ms of fragmentation 960(100% of CIR) or 912(95% of CIR) ?? Regards Claude -- This email was Anti Virus checked. Disclaimer The information in this Internet e-mail is confidential and may be legally privileged. It is intended solely for the addressee. Access to this Internet e-mail by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. When addressed to our clients any opinions or advice contained in this e-mail are subject to the terms and conditions expressed in our governing terms of business. * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Looking for vouchers for sale
Hi guys, Any vouchers for sale (only low price). Thanks a lot Naoufal -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Steffen Lehmann Sent: Saturday, February 05, 2011 4:26 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] proctorlabs vouchers for sale Hi all, i have 20 8h proctorlabs.com Lab vouchers to sell. Please send my a mail directly, if you´re interested. Do not need it anymore, got my number yesterday :-) Kind regards Steffen CCIE#28124 (Voice) ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] MVA DISA NOT WORKING
Dear Gents, I'm working on MVA scenario. I've setup the scenario for MVA and SNR. SNR is working well. MVA is working partially, meaning when I call the MVA number the system asking me for the pin then 1 to make a call. But when I pressed 1 and then the number # I heard only silence and then the call disconnected. Any Idea? Thanks a lot NAOUFAL KERBOUTE SERVICE MANAGER [Description: MHD_infotech] Post Box 880, Postal Code 112, Ruwi Sultanate of Oman Telephone 24 835 252, 24 834 848 24 834 400, 24 836 226 Mobile 9604 2593 E-Mail naou...@mhdinfotech.commailto:naou...@mhdinfotech.com Web www.mhdinfotech.comhttp://www.mhdinfotech.com/ Working hours 7:30 AM to 5:00 PM * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * inline: image001.png___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] graded labs remote controle
Dear Wayne, So now we can use Phoneview to control all phones lab? Many Thanks, Naoufal. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Wayne Lawson Sent: Monday, January 17, 2011 1:55 AM To: Wael Agina Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] graded labs remote controle Wael, Phoneview @ www.unifiedfx.comhttp://www.unifiedfx.com will enable users to remotely manage phones. This questions comes at a good time as we are preparing to make an announcement regarding this - very soon. Regards, Wayne A. Lawson II - CCIE #5244 (RS) Founder, President CEO - IPexpert, Inc., Proctor Labs, Inc. Platinum Solutions Group, LLC. Mailto: wlaw...@ipexpert.commailto:wlaw...@ipexpert.com Telephone: +1.810.334.1564 eFax: +1.810.454.0244 ::Message sent from iPhone IPexpert Proctor Labs are premier providers 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/communitieshttp://www.ipexpert.com/communities and our public website at www.ipexpert.comhttp://www.ipexpert.com or www.proctorlabs.comhttp://www.proctorlabs.com. CCIE-focused job community located at www.platinumplacementservices.comhttp://www.platinumplacementservices.com. Connect @ www.WayneLawson.comhttp://www.WayneLawson.com. On Jan 16, 2011, at 4:35 PM, Wael Agina waelag...@gmail.commailto:waelag...@gmail.com wrote: Hi Wayne, Do IPexpert going to have any phone remote control software , so we can fully utilize the proctorlabs and test all features ? I am just asking if you have any future plan regarding this matter. Many Thanks, Wael Agina On Sun, Jan 16, 2011 at 8:43 PM, Wayne Lawson groupst...@ipexpert.commailto:groupst...@ipexpert.com wrote: Um. Are you aware of our own racks @ www.ProctorLabs.comhttp://www.ProctorLabs.com? Regards, Wayne A. Lawson II - CCIE #5244 (RS) Founder, President CEO - IPexpert, Inc., Proctor Labs, Inc. Platinum Solutions Group, LLC. Mailto: wlaw...@ipexpert.commailto:wlaw...@ipexpert.com Telephone: +1.810.334.1564 eFax: +1.810.454.0244 ::Message sent from iPhone IPexpert Proctor Labs are premier providers 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/communitieshttp://www.ipexpert.com/communities and our public website at www.ipexpert.comhttp://www.ipexpert.com or www.proctorlabs.comhttp://www.proctorlabs.com. CCIE-focused job community located at www.platinumplacementservices.comhttp://www.platinumplacementservices.com. Connect @ www.WayneLawson.comhttp://www.WayneLawson.com. On Jan 16, 2011, at 11:35 AM, Joli-coeur Wouter jwou...@gmail.commailto:jwou...@gmail.com wrote: graded labs but i am starting to fell sorry about that Any other good labs out there? 3 days before exam and i need something very reliable. Graded lab isvery slow and some thing just don't work today On Sun, Jan 16, 2011 at 5:31 PM, Wayne Lawson groupst...@ipexpert.commailto:groupst...@ipexpert.com wrote: Are you using Graded Labs or Proctor Labs? If you're using Graded Labs - contact their support people. Regards, Wayne A. Lawson II - CCIE #5244 (RS) Founder, President CEO - IPexpert, Inc., Proctor Labs, Inc. Platinum Solutions Group, LLC. Mailto: wlaw...@ipexpert.commailto:wlaw...@ipexpert.com Telephone: +1.810.334.1564 eFax: +1.810.454.0244 ::Message sent from iPhone IPexpert Proctor Labs are premier providers 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/communitieshttp://www.ipexpert.com/communities and our public website at www.ipexpert.comhttp://www.ipexpert.com or www.proctorlabs.comhttp://www.proctorlabs.com. CCIE-focused job community located at www.platinumplacementservices.comhttp://www.platinumplacementservices.com. Connect @ www.WayneLawson.comhttp://www.WayneLawson.com. On Jan 16, 2011, at 10:44 AM, Joli-coeur Wouter jwou...@gmail.commailto:jwou...@gmail.com wrote: Hi, I am using gradedlabs to study, however is have problems using phone remote software to connect to CME maneged phones Wondering if anybody here was able to succesfully connect to CME manged phones using phone remote Don't understand what i am doing wrong i can
Re: [OSL | CCIE_Voice] proctorlabs vpn connection with ASA 5505
Hi guys, It is possible to use vpn client on the PC and use hardware phones? From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Julien Krieger Sent: Sunday, January 09, 2011 12:55 PM To: Carl Baccus Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] proctorlabs vpn connection with ASA 5505 Hi Carl, I had the same wish to use my ASA 5505 to connect to protorlab's vracks but proctorlab's support told me that they were not supporting it anymore as it wasn't working corretly. They told me to use one of the other available options. Julien 2011/1/9 Carl Baccus cjbac...@gmail.commailto:cjbac...@gmail.com I am currently working on connecting up my ASA at home for the proctorlabs voice vRack. I have read the article posted by Mark Snow, here on this forum: http://www.onlinestudylist.com/archives/ccie_voice/2009-September/010787.html and online elsewhere. I cannot find any actual config for the remaining ASA configuration, and even though I am able to establish a VPN connection to the Vrack, I cannot seem to route correctly to the 10.10.0.0/16http://10.10.0.0/16 address space contains the voice components. This is my configuration so far: ASA Version 7.2(2) ! hostname CarlASA domain-name .net enable password X names ! interface Vlan1 nameif inside security-level 100 ip address 192.168.1.1 255.255.255.0 ! interface Vlan2 nameif outside security-level 0 ip address dhcp ! interface Ethernet0/0 switchport access vlan 2 ! interface Ethernet0/1 ! interface Ethernet0/2 ! interface Ethernet0/3 ! interface Ethernet0/4 ! interface Ethernet0/5 ! interface Ethernet0/6 ! interface Ethernet0/7 ! passwd ftp mode passive dns server-group DefaultDNS domain-name X.net access-list inside_access_in extended permit ip any any log access-list outside_access_in extended permit ip any any log access-list outside_access_out extended permit ip any any log access-list inside_access_out extended permit ip any any log pager lines 24 logging enable logging asdm informational mtu inside 1500 mtu outside 1500 icmp unreachable rate-limit 1 burst-size 1 asdm image disk0:/asdm-522.bin no asdm history enable arp timeout 14400 access-group inside_access_in in interface inside access-group inside_access_out out interface inside access-group outside_access_in in interface outside access-group outside_access_out out interface outside route outside 0.0.0.0 0.0.0.0 172.16.3.1 1 timeout xlate 3:00:00 timeout conn 1:00:00 half-closed 0:10:00 udp 0:02:00 icmp 0:00:02 timeout sunrpc 0:10:00 h323 0:05:00 h225 1:00:00 mgcp 0:05:00 mgcp-pat 0:05:00 timeout sip 0:30:00 sip_media 0:02:00 sip-invite 0:03:00 sip-disconnect 0:02:00 timeout uauth 0:05:00 absolute username X password aaa authentication ssh console LOCAL http server enable http 192.168.1.0 255.255.255.0 inside no snmp-server location no snmp-server contact snmp-server enable traps snmp authentication linkup linkdown coldstart crypto isakmp policy 65535 authentication pre-share encryption 3des hash sha group 2 lifetime 86400 telnet timeout 5 ssh 0.0.0.0 0.0.0.0 inside ssh timeout 60 console timeout 0 dhcpd address 192.168.1.10-192.168.1.20 inside dhcpd dns 192.168.1.1 interface inside dhcpd option 3 ip 192.168.1.1 interface inside dhcpd option 150 ip 10.10.210.10 10.10.210.11 interface inside dhcpd enable inside ! vpnclient server 74.126.20.247 vpnclient mode network-extension-mode vpnclient vpngroup vpodgroup password vpnclient username password vpnclient ipsec-over-tcp port 80 vpnclient enable ! class-map inspection_default match default-inspection-traffic ! ! policy-map type inspect dns preset_dns_map parameters message-length maximum 512 policy-map global_policy class inspection_default inspect dns preset_dns_map inspect ftp inspect h323 h225 inspect h323 ras inspect netbios inspect rsh inspect rtsp inspect skinny inspect esmtp inspect sqlnet inspect sunrpc inspect tftp inspect sip inspect xdmcp ! service-policy global_policy global prompt hostname context I cannot seem to get any kind of route to the inside of the voice vRack though. I am not great at security or switching and routing, any help with this would be great. I will post a full tutorial with pictures once I get it figured out for all others to use! Thanks in advance! --Carl ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC.
Re: [OSL | CCIE_Voice] proctorlabs vpn connection with ASA 5505
Is there any other option? I have internet over 3G USB dongle, can I use another router for that other than cisco? -Original Message- From: dew.s...@gmail.com [mailto:dew.s...@gmail.com] Sent: Sunday, January 09, 2011 3:19 PM To: Julien Krieger; Carl Baccus; Naoufal Kerboute Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] proctorlabs vpn connection with ASA 5505 Nope. At least you need a cisco router to make a vpn connection in order to support hardware phones. -- Dew Swen / sent from mobile device CCVP, CCDP, CCNP-Original Message- From: Naoufal Kerboute Sent: 09/01/2011, 12:02 To: Julien Krieger; Carl Baccus Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] proctorlabs vpn connection with ASA 5505 Hi guys, It is possible to use vpn client on the PC and use hardware phones? From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Julien Krieger Sent: Sunday, January 09, 2011 12:55 PM To: Carl Baccus Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] proctorlabs vpn connection with ASA 5505 Hi Carl, I had the same wish to use my ASA 5505 to connect to protorlab's vracks but proctorlab's support told me that they were not supporting it anymore as it wasn't working corretly. They told me to use one of the other available options. Julien 2011/1/9 Carl Baccus cjbac...@gmail.commailto:cjbac...@gmail.com I am currently working on connecting up my ASA at home for the proctorlabs voice vRack. I have read the article posted by Mark Snow, here on this forum: http://www.onlinestudylist.com/archives/ccie_voice/2009-September/010787.html and online elsewhere. I cannot find any actual config for the remaining ASA configuration, and even though I am able to establish a VPN connection to the Vrack, I cannot seem to route correctly to the 10.10.0.0/16http://10.10.0.0/16 address space contains the voice components. This is my configuration so far: ASA Version 7.2(2) ! hostname CarlASA domain-name .net enable password X names ! interface Vlan1 nameif inside security-level 100 ip address 192.168.1.1 255.255.255.0 ! interface Vlan2 nameif outside security-level 0 ip address dhcp ! interface Ethernet0/0 switchport access vlan 2 ! interface Ethernet0/1 ! interface Ethernet0/2 ! interface Ethernet0/3 ! interface Ethernet0/4 ! interface Ethernet0/5 ! interface Ethernet0/6 ! interface Ethernet0/7 ! passwd ftp mode passive dns server-group DefaultDNS domain-name X.net access-list inside_access_in extended permit ip any any log access-list outside_access_in extended permit ip any any log access-list outside_access_out extended permit ip any any log access-list inside_access_out extended permit ip any any log pager lines 24 logging enable logging asdm informational mtu inside 1500 mtu outside 1500 icmp unreachable rate-limit 1 burst-size 1 asdm image disk0:/asdm-522.bin no asdm history enable arp timeout 14400 access-group inside_access_in in interface inside access-group inside_access_out out interface inside access-group outside_access_in in interface outside access-group outside_access_out out interface outside route outside 0.0.0.0 0.0.0.0 172.16.3.1 1 timeout xlate 3:00:00 timeout conn 1:00:00 half-closed 0:10:00 udp 0:02:00 icmp 0:00:02 timeout sunrpc 0:10:00 h323 0:05:00 h225 1:00:00 mgcp 0:05:00 mgcp-pat 0:05:00 timeout sip 0:30:00 sip_media 0:02:00 sip-invite 0:03:00 sip-disconnect 0:02:00 timeout uauth 0:05:00 absolute username X password aaa authentication ssh console LOCAL http server enable http 192.168.1.0 255.255.255.0 inside no snmp-server location no snmp-server contact snmp-server enable traps snmp authentication linkup linkdown coldstart crypto isakmp policy 65535 authentication pre-share encryption 3des hash sha group 2 lifetime 86400 telnet timeout 5 ssh 0.0.0.0 0.0.0.0 inside ssh timeout 60 console timeout 0 dhcpd address 192.168.1.10-192.168.1.20 inside dhcpd dns 192.168.1.1 interface inside dhcpd option 3 ip 192.168.1.1 interface inside dhcpd option 150 ip 10.10.210.10 10.10.210.11 interface inside dhcpd enable inside ! vpnclient server 74.126.20.247 vpnclient mode network-extension-mode vpnclient vpngroup vpodgroup password vpnclient username password vpnclient ipsec-over-tcp port 80 vpnclient enable ! class-map inspection_default match default-inspection-traffic ! ! policy-map type inspect dns preset_dns_map parameters message-length maximum 512 policy-map global_policy class inspection_default inspect dns preset_dns_map inspect ftp inspect h323 h225 inspect h323 ras inspect netbios inspect rsh inspect rtsp inspect skinny inspect esmtp inspect sqlnet inspect sunrpc inspect tftp inspect sip inspect xdmcp ! service-policy global_policy global prompt hostname context I cannot seem to get any kind of route to the inside of the voice vRack though. I am not great at security
Re: [OSL | CCIE_Voice] Gatekeeper trunk registration (SUB)
Hi, I had the same problem because of the replication. Try to reset your replication. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of manishankar pandey Sent: Monday, January 10, 2011 6:58 AM To: Joli-coeur Wouter; ccie_voice@onlinestudylist.com; bkvalent...@gmail.com Subject: Re: [OSL | CCIE_Voice] Gatekeeper trunk registration (SUB) Try Changing the CCM Group listing nd make the Sub the 1st in the listing or priority and see whether SUB trunk is getting registered?? Thanks M --- On Mon, 1/10/11, bkvalent...@gmail.commailto:bkvalent...@gmail.com bkvalent...@gmail.commailto:bkvalent...@gmail.com wrote: From: bkvalent...@gmail.commailto:bkvalent...@gmail.com bkvalent...@gmail.commailto:bkvalent...@gmail.com Subject: Re: [OSL | CCIE_Voice] Gatekeeper trunk registration (SUB) To: Joli-coeur Wouter jwou...@gmail.commailto:jwou...@gmail.com, ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Date: Monday, January 10, 2011, 5:40 AM Could be a replication problem. Make sue the ccm cluster is in sync. Sent from my Verizon Wireless Phone - Reply message - From: Joli-coeur Wouter jwou...@gmail.com Date: Sun, Jan 9, 2011 5:52 pm Subject: [OSL | CCIE_Voice] Gatekeeper trunk registration (SUB) To: ccie_voice@onlinestudylist.com Thanks for the info I have a sub configured for the correct CM group and the device pool has the correct CM group and the trunk is in the correct device pool. I think it might be a vmware problem or database problem? i just want to be sure that normally both trunks should show up if they are in the correct device group and the device group has the correct CM group. Regards, Joli-coeur Wouter On Sun, Jan 9, 2011 at 11:30 PM, Miron Kobelski findko...@gmail.commailto:findko...@gmail.com wrote: Hi, It seems you don't have Sub in CM Group configured for your trunk. Remember to restart it later. regards kobel On Sun, Jan 9, 2011 at 22:39, Joli-coeur Wouter jwou...@gmail.commailto:jwou...@gmail.comwrote: Hello, I am running into some problems with cucm/gatekeeper configuration. For some reason only the PUB trunk is being registered. Is the SUB trunk to register automatically or do i need to configure something for that to happen ? interface Loopback0 ip address 177.1.254.1 255.255.255.255 ip pim dense-mode h323-gateway voip interface h323-gateway voip bind srcaddr 177.1.254.1 gatekeeper zone local GK cisco.com 177.1.254.1 zone prefix GK 2* zone prefix GK 4* no shutdown ! I created a gatekeeper in CUCM (pub) andi created a H.225 Trunk (Gatekeeper Controlled trunk) I added the trunk name GK-Trunk in the Device Name of GK-controlled Trunk That Will Use Port 1720 service Am i missing something ? PUB/SUB are vmware. Thanks for your help GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags --- - --- - - - *177.1.10.10 1720 177.1.10.10 32816 GKVOIP-GW *H323-ID: GK-Trunk_1 Voice Capacity Max.= Avail.= Current.= 0 177.1.254.2 1720 177.1.254.2 61105 GKH323-GW Voice Capacity Max.= Avail.= Current.= 0 177.1.254.3 1720 177.1.254.3 61431 GKVOIP-GW H323-ID: CUCME E164-ID: 4001 E164-ID: 4002 E164-ID: 4003 E164-ID: 4300 E164-ID: 4301 E164-ID: 4302 Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 3 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com -Inline Attachment Follows- ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. *
[OSL | CCIE_Voice] CUCM Huntgroup (No Fwd)
Hi Gents, I'm working on Hunt group configuration on CUCM. I wanted the CallFwdNoAn and CallFwdBusy settings shoud be honored for the user who forwarded the call instead of the Hunt pilot/ Below My config: One Hunt Group One Hunt List One Hunt Pilot (5000) and I've checked Use Personal preferences for CallFwdNoAn and CallFwdBusy The problem is whenever I receive a call on the hunt Pilot the same ring in circular fashion but the CallFwdNoAn and CallFwdBusy doesn't happened even the forward settings has been set on the members of the hunt group. Simply I heard busy tone without forward. Any Ideas? Regards, Naoufal * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing
Have you tried what I told you? -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Hough, Earl Sent: Saturday, January 01, 2011 3:53 PM To: John Nield; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing This scenario doesn't involve CUCM. It was only two CME sites - BR2 and BR1 using the HQ-RTR as a GK/CUBE routing and media termination resource. -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of John Nield Sent: Saturday, January 01, 2011 5:28 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing On 1/01/2011 3:37 PM, ccie_voice-requ...@onlinestudylist.com wrote: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing Hi sounds like you br1 xcoders is not being allocated to the call, therefore you have no MTP function and a world of hurt. this theory assumes you're HQ phones are ok. to verify this run up rtmt and check the number of xcoding resources being used, also check if the graph for unable to allocate resources. i assume your br1 MRLG contains a hardware xcoder that would be used for this instance, a software MTP in my experience fails due to the g729 codec the br1 sites will be requesting. good luck. regards john ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ The information contained in this transmission is confidential. It is intended solely for the use of the individual(s) or organization(s) to whom it is addressed. Any disclosure, copying or further distribution is not permitted unless such privilege is explicitly granted in writing by PC Mall, Inc. Furthermore, PC Mall, Inc. is not responsible for the proper and complete transmission of the substance of this communication, nor for any delay in its receipt. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing
I sent an email, check below: Voice service voip H323 H225 connect-passthru ! If doesn't work please try Voice service voip H323 h225 start-h245 on-connect Hope this will help. If not try to configure MTP. Regards, Naoufal, -Original Message- From: Hough, Earl [mailto:earl.ho...@pcmallservices.com] Sent: Saturday, January 01, 2011 4:29 PM To: Naoufal Kerboute Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing What am I suppose to try? You're suggestions involve CUCM, which this scenario isn't using. When I get a chance later today, I will try throwing a hardware transcoder on BR1. -Original Message- From: Naoufal Kerboute [mailto:naou...@mhdinfotech.com] Sent: Saturday, January 01, 2011 7:13 AM To: Hough, Earl Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing Have you tried what I told you? -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Hough, Earl Sent: Saturday, January 01, 2011 3:53 PM To: John Nield; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing This scenario doesn't involve CUCM. It was only two CME sites - BR2 and BR1 using the HQ-RTR as a GK/CUBE routing and media termination resource. -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of John Nield Sent: Saturday, January 01, 2011 5:28 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing On 1/01/2011 3:37 PM, ccie_voice-requ...@onlinestudylist.com wrote: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing Hi sounds like you br1 xcoders is not being allocated to the call, therefore you have no MTP function and a world of hurt. this theory assumes you're HQ phones are ok. to verify this run up rtmt and check the number of xcoding resources being used, also check if the graph for unable to allocate resources. i assume your br1 MRLG contains a hardware xcoder that would be used for this instance, a software MTP in my experience fails due to the g729 codec the br1 sites will be requesting. good luck. regards john ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ The information contained in this transmission is confidential. It is intended solely for the use of the individual(s) or organization(s) to whom it is addressed. Any disclosure, copying or further distribution is not permitted unless such privilege is explicitly granted in writing by PC Mall, Inc. Furthermore, PC Mall, Inc. is not responsible for the proper and complete transmission of the substance of this communication, nor for any delay in its receipt. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ The information contained in this transmission is confidential. It is intended solely for the use of the individual(s) or organization(s) to whom it is addressed. Any disclosure, copying or further distribution is not permitted unless such privilege is explicitly granted in writing by PC Mall, Inc. Furthermore, PC Mall, Inc. is not responsible for the proper and complete transmission of the substance of this communication, nor for any delay in its receipt
Re: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing
Uncheck Wait for Far End H.245 Terminal Capability Set on the GK trunk on CUCM. Naoufal From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Hough, Earl Sent: Friday, December 31, 2010 6:04 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing Everyone, Been struggling with a scenario which seems to be rock-solid for SCCP endpoints. The topology I'm working with is as follows: BR2 (UCME 7.0) G.729r8-- HQ (GK/CUBE) --G.711ulaw-- BR2 (UCME 7.0) What I've been trying to do is to connect the two remote branches as H323 gateways utilizing the GK/CUBE resources of the HQ router to provide GK address resolution and media termination. I have no problems getting this to work using viazone processing in both directions when using exclusively SCCP endpoints. What I have a problem with is SIP endpoints on BR2 being able to call SCCP endpoints at BR1. The SIP endpoints at BR2 will initiate a call to a SCCP phone at BR1 and the BR1 phone will ring, but after the call is answered on the BR1 side, the BR2 side continues to ring out and never completes the call. If I hang up the call from the BR1 side both sides disconnect, so it appears as though signaling is still working, just not the media path. It also appears as though the H245 capability set is never completed when a SIP endpoint at BR2 initiates a call to BR1. It does correctly work when an SCCP endpoint at BR2 initiates a call to BR1. I've been scratching my head looking over debugs and traces for several hours here and though I'd throw it out to the group as to what anyone's thoughts would be as to why this isn't working correctly. If I go straight through from BR2 to BR1 only using GK address resolution and not via-zone processing in that direction, the SIP endpoints are able to complete calls. Any thoughts on this from group? The relevant config portions are as follows: HQ-RTR (GK/CUBE) --- ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip ! ! voice-card 0 no dspfarm dsp services dspfarm ! ! interface Loopback0 ip address 10.10.110.1 255.255.255.0 h323-gateway voip interface h323-gateway voip id CUBE ipaddr 10.10.110.1 1719 h323-gateway voip h323-id HQ-RTR h323-gateway voip bind srcaddr 10.10.110.1 ! ! sccp local FastEthernet0/0.20 sccp ccm 10.10.200.3 identifier 1 version 6.0 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register XCD-CME ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 4 associate application SCCP ! ! dial-peer voice 3000 voip incoming called-number 3...$ dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 4000 voip destination-pattern 3...$ session target ras dtmf-relay h245-alphanumeric no vad ! dial-peer voice 3100 voip incoming called-number 1...$ dtmf-relay h245-alphanumeric no vad ! dial-peer voice 4100 voip destination-pattern 1...$ session target ras dtmf-relay h245-alphanumeric codec g711ulaw no vad ! ! gateway timer receive-rtp 1200 ! ! ! gatekeeper zone local BR1 cisco.com 10.10.110.1 outvia CUBE zone local CUBE cisco.com zone local BR2 cisco.com outvia CUBE zone prefix BR1 1... zone prefix BR2 3... gw-type-prefix 1#* default-technology no shutdown ! ! telephony-service sdspfarm units 1 sdspfarm transcode sessions 4 sdspfarm tag 1 XCD-CME load 7961 SCCP41.8-3-3S load 7962 SCCP42.8-3-3S load 7965 SCCP45.8-3-3S max-ephones 10 max-dn 20 no-reg both ip source-address 10.10.200.3 port 2000 max-conferences 8 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Dec 30 2010 07:46:37 ! BR1 (CME w/ SCCP-only endpoints) - ! interface Loopback0 ip address 10.10.110.2 255.255.255.255 ip ospf network point-to-point h323-gateway voip interface h323-gateway voip id BR1 ipaddr 10.10.110.1 1719 h323-gateway voip h323-id BR1-RTR h323-gateway voip tech-prefix 1# h323-gateway voip bind srcaddr 10.10.110.2 ! ! ! ! ! dial-peer voice 4000 voip destination-pattern 3...$ session target ras dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 3000 voip incoming called-number 1...$ dtmf-relay h245-alphanumeric codec g711ulaw no vad ! BR2 (CME w/ SCCP and SIP endpoints) ! ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip sip bind control source-interface vlan400 bind media source-interface vlan400 registrar server ! ! voice register global mode cme source-address 10.10.202.1 port 5060 max-dn 20 max-pool 10 load 7971 SIP70.8-3-3S load 7965 SIP45.8-3-3S load 7961 SIP41.8-3-3S authenticate register timezone 42 time-format 24 date-format
Re: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing
Ah sorry I didn't get it. Please try Voice service voip H323 H225 connect-passthru ! If doesn't work please try Voice service voip H323 h225 start-h245 on-connect Hope this will help. Regards, Naoufal, From: Hough, Earl [mailto:earl.ho...@pcmallservices.com] Sent: Saturday, January 01, 2011 9:14 AM To: Naoufal Kerboute; ccie_voice@onlinestudylist.com Subject: RE: CME SIP Endpoints with GK/CUBE Routing This scenario doesn't involve Callmanager. It was two CME sites using HQ GK/CUBE to connect the two sites. From: Naoufal Kerboute [mailto:naou...@mhdinfotech.com] Sent: Friday, December 31, 2010 11:35 PM To: Hough, Earl; ccie_voice@onlinestudylist.com Subject: RE: CME SIP Endpoints with GK/CUBE Routing Uncheck Wait for Far End H.245 Terminal Capability Set on the GK trunk on CUCM. Naoufal From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Hough, Earl Sent: Friday, December 31, 2010 6:04 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing Everyone, Been struggling with a scenario which seems to be rock-solid for SCCP endpoints. The topology I'm working with is as follows: BR2 (UCME 7.0) G.729r8-- HQ (GK/CUBE) --G.711ulaw-- BR2 (UCME 7.0) What I've been trying to do is to connect the two remote branches as H323 gateways utilizing the GK/CUBE resources of the HQ router to provide GK address resolution and media termination. I have no problems getting this to work using viazone processing in both directions when using exclusively SCCP endpoints. What I have a problem with is SIP endpoints on BR2 being able to call SCCP endpoints at BR1. The SIP endpoints at BR2 will initiate a call to a SCCP phone at BR1 and the BR1 phone will ring, but after the call is answered on the BR1 side, the BR2 side continues to ring out and never completes the call. If I hang up the call from the BR1 side both sides disconnect, so it appears as though signaling is still working, just not the media path. It also appears as though the H245 capability set is never completed when a SIP endpoint at BR2 initiates a call to BR1. It does correctly work when an SCCP endpoint at BR2 initiates a call to BR1. I've been scratching my head looking over debugs and traces for several hours here and though I'd throw it out to the group as to what anyone's thoughts would be as to why this isn't working correctly. If I go straight through from BR2 to BR1 only using GK address resolution and not via-zone processing in that direction, the SIP endpoints are able to complete calls. Any thoughts on this from group? The relevant config portions are as follows: HQ-RTR (GK/CUBE) --- ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip ! ! voice-card 0 no dspfarm dsp services dspfarm ! ! interface Loopback0 ip address 10.10.110.1 255.255.255.0 h323-gateway voip interface h323-gateway voip id CUBE ipaddr 10.10.110.1 1719 h323-gateway voip h323-id HQ-RTR h323-gateway voip bind srcaddr 10.10.110.1 ! ! sccp local FastEthernet0/0.20 sccp ccm 10.10.200.3 identifier 1 version 6.0 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register XCD-CME ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 4 associate application SCCP ! ! dial-peer voice 3000 voip incoming called-number 3...$ dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 4000 voip destination-pattern 3...$ session target ras dtmf-relay h245-alphanumeric no vad ! dial-peer voice 3100 voip incoming called-number 1...$ dtmf-relay h245-alphanumeric no vad ! dial-peer voice 4100 voip destination-pattern 1...$ session target ras dtmf-relay h245-alphanumeric codec g711ulaw no vad ! ! gateway timer receive-rtp 1200 ! ! ! gatekeeper zone local BR1 cisco.com 10.10.110.1 outvia CUBE zone local CUBE cisco.com zone local BR2 cisco.com outvia CUBE zone prefix BR1 1... zone prefix BR2 3... gw-type-prefix 1#* default-technology no shutdown ! ! telephony-service sdspfarm units 1 sdspfarm transcode sessions 4 sdspfarm tag 1 XCD-CME load 7961 SCCP41.8-3-3S load 7962 SCCP42.8-3-3S load 7965 SCCP45.8-3-3S max-ephones 10 max-dn 20 no-reg both ip source-address 10.10.200.3 port 2000 max-conferences 8 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Dec 30 2010 07:46:37 ! BR1 (CME w/ SCCP-only endpoints) - ! interface Loopback0 ip address 10.10.110.2 255.255.255.255 ip ospf network point-to-point h323-gateway voip interface h323-gateway voip id BR1 ipaddr 10.10.110.1 1719 h323-gateway voip h323-id BR1-RTR h323-gateway voip tech-prefix 1# h323-gateway voip bind srcaddr 10.10.110.2 ! ! ! ! ! dial-peer voice 4000 voip destination-pattern 3...$ session target ras
[OSL | CCIE_Voice] SRST (Incoming call not working)
Hi Gents, I'm working on SRST LAB and I have a small problem. I'm unable to receive calls. Below the config: ccm-manager switchback immediate ccm-manager fallback-mgcp ccm-manager redundant-host 10.10.210.10 ccm-manager mgcp ! mgcp mgcp call-agent 10.10.210.11 service-type mgcp version 0.1 mgcp bind control source-interface Loopback0 mgcp bind media source-interface Loopback0 ! mgcp profile default ! ! call-manager-fallback max-conferences 8 gain -6 transfer-system full-consult ip source-address 10.10.202.1 port 2000 max-ephones 5 max-dn 30 dual-line dialplan-pattern 1 617863 extension-length 4 transfer-pattern T voicemail 4200 call-forward busy 4200 call-forward noan 4200 timeout 20 mwi relay moh music-on-hold.au time-format 24 date-format dd-mm-yy Below the debug isdn q931: BR1-SRST# *Dec 29 13:29:54.580: ISDN Se0/2/0:23 Q931: RX - SETUP pd = 8 callref = 0x00CA Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Progress Ind i = 0x8583 - Origination address is non-ISDN Calling Party Number i = 0x4180, '8632683' Plan:ISDN, Type:Subscriber(local) Called Party Number i = 0xA1, '6178631002' Plan:ISDN, Type:National *Dec 29 13:29:54.612: ISDN Se0/2/0:23 Q931: TX - CALL_PROC pd = 8 callref = 0x80CA Channel ID i = 0xA98381 Exclusive, Channel 1 *Dec 29 13:29:54.616: ISDN Se0/2/0:23 Q931: TX - DISCONNECT pd = 8 callref = 0x80CA Cause i = 0x829B - Destination out of order *Dec 29 13:29:54.620: ISDN Se0/2/0:23 Q931: RX - RELEASE pd = 8 callref = 0x00CA *Dec 29 13:29:54.676: ISDN Se0/2/0:23 Q931: TX - RELEASE_COMP pd = 8 callref = 0x80CA Below the show dial-peer voice summary BR1-SRST#sh dial-peer voice summary dial-peer hunt 0 ADPRE PASSOUT TAGTYPE MIN OPER PREFIXDEST-PATTERN FER THRU SESS-TARGETSTAT PORT 20037 pots up up 1001$ 0 50/0/1 20038 pots up up 1002$ 0 50/0/2 20039 pots up down 0 50/0/3 20040 pots up down 0 50/0/4 20041 pots up down 0 50/0/5 20042 pots up down 0 50/0/6 20043 pots up down 0 50/0/7 20044 pots up down 0 50/0/8 20045 pots up down 0 50/0/9 20046 pots up down 0 50/0/10 20047 pots up up 6178631001$0 50/0/1 20048 pots up up 6178631002$0 50/0/2 Show ephone registred: BR1-SRST#sh ephone registered ephone-1[0] Mac:0026.0BD6.CBB7 TCP socket:[2] activeLine:0 whisperLine:0 REGISTERED in SCCP ver 17/12 max_streams=5 mediaActive:0 whisper_mediaActive:0 startMedia:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:9 IP:142.102.65.79 25596 7965 keepalive 57 max_line 6 available_line 2 button 1: dn 1 number 1001 CM Fallback CH1 IDLE CH2 IDLE mwi Preferred Codec: g711ulaw ephone-2[1] Mac:081F.F363.B37D TCP socket:[1] activeLine:0 whisperLine:0 REGISTERED in SCCP ver 17/12 max_streams=5 mediaActive:0 whisper_mediaActive:0 startMedia:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:9 IP:142.102.65.78 14399 7945 keepalive 58 max_line 2 available_line 2 button 1: dn 2 number 1002 CM Fallback CH1 IDLE CH2 IDLE Preferred Codec: g711ulaw Please advise. Thank you, Naoufal * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. *
Re: [OSL | CCIE_Voice] SRST (Incoming call not working)
Please ignore my email. I forgot the redundant host (was online) :S From: Naoufal Kerboute Sent: Wednesday, December 29, 2010 5:37 PM To: ccie_voice@onlinestudylist.com Subject: SRST (Incoming call not working) Hi Gents, I'm working on SRST LAB and I have a small problem. I'm unable to receive calls. Below the config: ccm-manager switchback immediate ccm-manager fallback-mgcp ccm-manager redundant-host 10.10.210.10 ccm-manager mgcp ! mgcp mgcp call-agent 10.10.210.11 service-type mgcp version 0.1 mgcp bind control source-interface Loopback0 mgcp bind media source-interface Loopback0 ! mgcp profile default ! ! call-manager-fallback max-conferences 8 gain -6 transfer-system full-consult ip source-address 10.10.202.1 port 2000 max-ephones 5 max-dn 30 dual-line dialplan-pattern 1 617863 extension-length 4 transfer-pattern T voicemail 4200 call-forward busy 4200 call-forward noan 4200 timeout 20 mwi relay moh music-on-hold.au time-format 24 date-format dd-mm-yy Below the debug isdn q931: BR1-SRST# *Dec 29 13:29:54.580: ISDN Se0/2/0:23 Q931: RX - SETUP pd = 8 callref = 0x00CA Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Progress Ind i = 0x8583 - Origination address is non-ISDN Calling Party Number i = 0x4180, '8632683' Plan:ISDN, Type:Subscriber(local) Called Party Number i = 0xA1, '6178631002' Plan:ISDN, Type:National *Dec 29 13:29:54.612: ISDN Se0/2/0:23 Q931: TX - CALL_PROC pd = 8 callref = 0x80CA Channel ID i = 0xA98381 Exclusive, Channel 1 *Dec 29 13:29:54.616: ISDN Se0/2/0:23 Q931: TX - DISCONNECT pd = 8 callref = 0x80CA Cause i = 0x829B - Destination out of order *Dec 29 13:29:54.620: ISDN Se0/2/0:23 Q931: RX - RELEASE pd = 8 callref = 0x00CA *Dec 29 13:29:54.676: ISDN Se0/2/0:23 Q931: TX - RELEASE_COMP pd = 8 callref = 0x80CA Below the show dial-peer voice summary BR1-SRST#sh dial-peer voice summary dial-peer hunt 0 ADPRE PASSOUT TAGTYPE MIN OPER PREFIXDEST-PATTERN FER THRU SESS-TARGETSTAT PORT 20037 pots up up 1001$ 0 50/0/1 20038 pots up up 1002$ 0 50/0/2 20039 pots up down 0 50/0/3 20040 pots up down 0 50/0/4 20041 pots up down 0 50/0/5 20042 pots up down 0 50/0/6 20043 pots up down 0 50/0/7 20044 pots up down 0 50/0/8 20045 pots up down 0 50/0/9 20046 pots up down 0 50/0/10 20047 pots up up 6178631001$0 50/0/1 20048 pots up up 6178631002$0 50/0/2 Show ephone registred: BR1-SRST#sh ephone registered ephone-1[0] Mac:0026.0BD6.CBB7 TCP socket:[2] activeLine:0 whisperLine:0 REGISTERED in SCCP ver 17/12 max_streams=5 mediaActive:0 whisper_mediaActive:0 startMedia:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:9 IP:142.102.65.79 25596 7965 keepalive 57 max_line 6 available_line 2 button 1: dn 1 number 1001 CM Fallback CH1 IDLE CH2 IDLE mwi Preferred Codec: g711ulaw ephone-2[1] Mac:081F.F363.B37D TCP socket:[1] activeLine:0 whisperLine:0 REGISTERED in SCCP ver 17/12 max_streams=5 mediaActive:0 whisper_mediaActive:0 startMedia:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:9 IP:142.102.65.78 14399 7945 keepalive 58 max_line 2 available_line 2 button 1: dn 2 number 1002 CM Fallback CH1 IDLE CH2 IDLE Preferred Codec: g711ulaw Please advise. Thank you, Naoufal * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any
Re: [OSL | CCIE_Voice] show gatekeeper gw
Check the device pool assigned to the trunk. You shoud have a device pool that contain the cucm group with sub first then pub. Naoufal From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of study2b ccie Sent: Thursday, December 30, 2010 6:50 AM To: OSL Subject: [OSL | CCIE_Voice] show gatekeeper gw Hello experts, when entering sh gatek gw I have: GATEWAY TYPE PREFIX TABLE = Prefix: 34* Zone GK master gateway list: 10.10.202.1:1720http://10.10.202.1:1720 CUCME Prefix: 1* Zone GK master gateway list: 10.10.210.10:1720http://10.10.210.10:1720 GK-Trunk_1 10.10.210.11:1720http://10.10.210.11:1720 GK-Trunk_2 I followed solution to the dot, and I still did not have HQ-RTR#sh gatek gw GATEWAY TYPE PREFIX TABLE = Prefix: 34* Zone GK master gateway list: 10.10.202.1:1720http://10.10.202.1:1720 CUCME Prefix: 1* Zone GK master gateway list: 10.10.210.10:1720http://10.10.210.10:1720 GK-Trunk_2 10.10.210.11:1720http://10.10.210.11:1720 GK-Trunk_1 It would show GK-Trunk_2 on top every time I restart the trunk, but it would become GK-Trunk_1 on top soon after. How do you guys do it to keep GK-Trunk_2 on the top constantly? Thank you in advance. * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] TEST
* * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] RE : Lab 5 Volume 2 LLQ sizing and RSVP CAC
Hi, I have a question, why proctor on lab 3 Vol 2 with MLP LFI + LLQ +FRST take into account the RSVP 1st call in the PQ? (mentioned in the question) Regards Naoufal De : ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] de la part de kobel [findko...@gmail.com] Date d'envoi : dimanche 27 juin 2010 17:12 À : Daniel Berlinski Cc : osl osl Objet : Re: [OSL | CCIE_Voice] Lab 5 Volume 2 LLQ sizing and RSVP CAC Hi, When calculating bandwidth needed for PQ, I wouldn't take into account the value used for initial call by RSVP. It's never actually used, it's only a worst case scenario. voice packets are small and are never fragmented by FRF.12. that's why additional 4B in header are not needed. On Sun, Jun 27, 2010 at 6:50 PM, Daniel Berlinski dberlin...@gmail.commailto:dberlin...@gmail.com wrote: Hello list Volume 2 lab 5 has a scenario asking us to allow for 4 concurrent g729 calls over Frame FRF.12 LFI using RSVP for CAC. Proctor Guide has calculated the size of the priority queue without taking into account that first call prior to capabilities exchange that RSVP negotiates at 40Kbps. In addition Proctor Guide has used Frame Relay payload of 4 Bytes instead of 8 Bytes for FR with LFI. I answered this question as follows: For 4 g729r8 concurrent calls over the WAN using RSVP for CAC: compressed ip/udp/rtp=2bytes FRF.12=8Bytes g729 payload @ 20ms=20Bytes 30*50*8/1000=12Kbps per call so 3 calls=36Kbps 1 call all @ worse case scenario compressed ip/udp/rtp=2bytes FRF.12=8Bytes g729 payload @ 10ms=10bytes 20*100*8/1000 = 1 call 16Kbps So 4 calls=36kbps + 16Kbps= 52Kbps configured in priority queue Can anyone let me know if my approach is right or wrong and if wrong why? Thanks a lot Daniel ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Multicast Moh from Flash of BR1
Hi, I'm playing with multicast moh over the flash of router, the configuration looks good and I can see in the debug the moh traffic, but on the PSTN phone I heard only bips. below the output of debug ephone moh: Jun 25 18:03:04.729: MoH route If Vlan240 ETHERNET 10.10.201.1 via ARP Jun 25 18:03:04.729: MoH route If Loopback0 46 10.10.110.2 via 10.10.110.2 Jun 25 18:03:09.173: ifs_read flash:music-on-hold.au end of file at 492545 read 3976 = 496521 Jun 25 18:03:09.177: moh tail fill from 24 at 0x4A0E9FF8 length 4024 Jun 25 18:03:10.017: MoH route If Vlan240 ETHERNET 10.10.201.1 via ARP Jun 25 18:03:10.017: MoH route If Loopback0 46 10.10.110.2 via 10.10.110.2 I've created a region who use g711 with other regions, activated the multicast on the source file and the moh server. Any Ideas? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] RE : Multicast Moh from Flash of BR1
Thanks Mark, However I'm not missing this configuration, may be the file is corrupted. Because I did it with the same way in lab1 vol1 on PL vRack, and Now I'm on Vol2Lab2. De : Mark Holloway [...@markholloway.com] Date d'envoi : vendredi 25 juin 2010 18:47 À : naoufal kerboute Cc : ccie_voice@onlinestudylist.com Objet : Re: [OSL | CCIE_Voice] Multicast Moh from Flash of BR1 Amy and Jeff helped me with this the other day. Here is what I was missing (from Jeff's reply) max-ephones 1 max-dn 1 ip source ip address of voice vlan/loop Also, under global config add the following command: ccm-manager music-on-hold bind source ip address configured under ccm-manager-fallback On Jun 25, 2010, at 12:24 PM, naoufal kerboute wrote: Hi, I'm playing with multicast moh over the flash of router, the configuration looks good and I can see in the debug the moh traffic, but on the PSTN phone I heard only bips. below the output of debug ephone moh: Jun 25 18:03:04.729: MoH route If Vlan240 ETHERNET 10.10.201.1 via ARP Jun 25 18:03:04.729: MoH route If Loopback0 46 10.10.110.2 via 10.10.110.2 Jun 25 18:03:09.173: ifs_read flash:music-on-hold.au end of file at 492545 read 3976 = 496521 Jun 25 18:03:09.177: moh tail fill from 24 at 0x4A0E9FF8 length 4024 Jun 25 18:03:10.017: MoH route If Vlan240 ETHERNET 10.10.201.1 via ARP Jun 25 18:03:10.017: MoH route If Loopback0 46 10.10.110.2 via 10.10.110.2 I've created a region who use g711 with other regions, activated the multicast on the source file and the moh server. Any Ideas? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] RE : Multicast Moh from Flash of BR1
Hi Ashar, I need to play in case to forget the stress of the lab, it's coming soon. I think you are right the problem on the UCM side, I will try to restart the Media voice app and back to you. The output of show ccm-manager music-on-hold BR1-RTR#show ccm-manager music-on-hold Current active multicast sessions : 0 So ucm didn't provide any multicast to the router. Regards De : Ashar Siddiqui [siddas...@gmail.com] Date d'envoi : vendredi 25 juin 2010 18:51 À : naoufal kerboute Cc : ccie_voice@onlinestudylist.com Objet : Re: [OSL | CCIE_Voice] Multicast Moh from Flash of BR1 First of all, why are you playing? You should be labbing properly as voice field is not a playground.. Joke aside... ;) What is your ‘show ccm-manager music-on-hold ? If no MoH streams are shown by this command then CCM has failed to provide the gateway with MoH Also keep in mind Tone on hold means there is a CM misconfiguration. Silence means the RTP is not getting to the router. Your case is CCM configuration issue. Check configuration again properly. Check at Pub CCM the MOH server - region/Device pool selected properly, reset it and then login to subscriber..go to the same place and reset it from there as well. Also restart Media Voice app. The output you posted means that you have proper telephony-service configuration (max-dn/max-ephone etc)...problem is at cucm side.. You can now continue playing.. Ash naoufal kerboute wrote: Hi, I'm playing with multicast moh over the flash of router, the configuration looks good and I can see in the debug the moh traffic, but on the PSTN phone I heard only bips. below the output of debug ephone moh: Jun 25 18:03:04.729: MoH route If Vlan240 ETHERNET 10.10.201.1 via ARP Jun 25 18:03:04.729: MoH route If Loopback0 46 10.10.110.2 via 10.10.110.2 Jun 25 18:03:09.173: ifs_read flash:music-on-hold.au end of file at 492545 read 3976 = 496521 Jun 25 18:03:09.177: moh tail fill from 24 at 0x4A0E9FF8 length 4024 Jun 25 18:03:10.017: MoH route If Vlan240 ETHERNET 10.10.201.1 via ARP Jun 25 18:03:10.017: MoH route If Loopback0 46 10.10.110.2 via 10.10.110.2 I've created a region who use g711 with other regions, activated the multicast on the source file and the moh server. Any Ideas? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] RE : Multicast Moh from Flash of BR1
Hi Ashar, I've tried what u told me but without success, also I've noticed that the call disconnected after 26 seconds below the debug for ccm-manager music-on-hold all BR1-RTR#debug ccm-manager music-on-hold all Call Manager music-on-hold all debugging is on BR1-RTR# Jun 25 19:06:01.529: moh_update_rtp: callID 38 dstCallID -1 Jun 25 19:06:01.537: moh_update_rtp: callID 38 dstCallID 37 Jun 25 19:06:01.537: moh_update_rtp: callID 38 dstCallID 37 BR1-RTR# Jun 25 19:06:03.885: moh_update_rtp: callID 38 dstCallID 37 Jun 25 19:06:03.885: moh_process_ccb: dstadr 192.168.10.22, callid 37, port 25612, codec 12, moh_en 0, moh_addr 0.0.0.0 Jun 25 19:06:03.889: moh_update_rtp: callID 38 dstCallID 37 BR1-RTR# Jun 25 19:06:06.389: moh_update_rtp: callID 38 dstCallID 37 BR1-RTR# Jun 25 19:06:31.393: moh_update_rtp: callID 38 dstCallID 37 Jun 25 19:06:31.409: moh_delete_ccb: called dstadr 0.0.0.0, callid 0 Any ideas? May be the QoS?? Thank you De : Ashar Siddiqui [siddas...@gmail.com] Date d'envoi : vendredi 25 juin 2010 18:51 À : naoufal kerboute Cc : ccie_voice@onlinestudylist.com Objet : Re: [OSL | CCIE_Voice] Multicast Moh from Flash of BR1 First of all, why are you playing? You should be labbing properly as voice field is not a playground.. Joke aside... ;) What is your ‘show ccm-manager music-on-hold ? If no MoH streams are shown by this command then CCM has failed to provide the gateway with MoH Also keep in mind Tone on hold means there is a CM misconfiguration. Silence means the RTP is not getting to the router. Your case is CCM configuration issue. Check configuration again properly. Check at Pub CCM the MOH server - region/Device pool selected properly, reset it and then login to subscriber..go to the same place and reset it from there as well. Also restart Media Voice app. The output you posted means that you have proper telephony-service configuration (max-dn/max-ephone etc)...problem is at cucm side.. You can now continue playing.. Ash naoufal kerboute wrote: Hi, I'm playing with multicast moh over the flash of router, the configuration looks good and I can see in the debug the moh traffic, but on the PSTN phone I heard only bips. below the output of debug ephone moh: Jun 25 18:03:04.729: MoH route If Vlan240 ETHERNET 10.10.201.1 via ARP Jun 25 18:03:04.729: MoH route If Loopback0 46 10.10.110.2 via 10.10.110.2 Jun 25 18:03:09.173: ifs_read flash:music-on-hold.au end of file at 492545 read 3976 = 496521 Jun 25 18:03:09.177: moh tail fill from 24 at 0x4A0E9FF8 length 4024 Jun 25 18:03:10.017: MoH route If Vlan240 ETHERNET 10.10.201.1 via ARP Jun 25 18:03:10.017: MoH route If Loopback0 46 10.10.110.2 via 10.10.110.2 I've created a region who use g711 with other regions, activated the multicast on the source file and the moh server. Any Ideas? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] RE : RE : Multicast Moh from Flash of BR1
GOT IT BR1-RTR#sh ccm-manager music-on-hold Current active multicast sessions : 1 Multicast RTP port Packets Call CodecIncoming Address number in/outid Interface === 239.1.1.1 16384 98/9849 g711ulaw Lo0 I forgot to do a no mgcp / mgcp on the BR1 router :) I've added the MRGL to the DP BR1 and it's associated to the gateway. so I had to restart the mgcp to take effect. Thank you guys De : ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] de la part de naoufal kerboute [naoufal.kerbo...@cbi.ma] Date d'envoi : vendredi 25 juin 2010 20:07 À : Ashar Siddiqui Cc : ccie_voice@onlinestudylist.com Objet : [OSL | CCIE_Voice] RE : Multicast Moh from Flash of BR1 Hi Ashar, I've tried what u told me but without success, also I've noticed that the call disconnected after 26 seconds below the debug for ccm-manager music-on-hold all BR1-RTR#debug ccm-manager music-on-hold all Call Manager music-on-hold all debugging is on BR1-RTR# Jun 25 19:06:01.529: moh_update_rtp: callID 38 dstCallID -1 Jun 25 19:06:01.537: moh_update_rtp: callID 38 dstCallID 37 Jun 25 19:06:01.537: moh_update_rtp: callID 38 dstCallID 37 BR1-RTR# Jun 25 19:06:03.885: moh_update_rtp: callID 38 dstCallID 37 Jun 25 19:06:03.885: moh_process_ccb: dstadr 192.168.10.22, callid 37, port 25612, codec 12, moh_en 0, moh_addr 0.0.0.0 Jun 25 19:06:03.889: moh_update_rtp: callID 38 dstCallID 37 BR1-RTR# Jun 25 19:06:06.389: moh_update_rtp: callID 38 dstCallID 37 BR1-RTR# Jun 25 19:06:31.393: moh_update_rtp: callID 38 dstCallID 37 Jun 25 19:06:31.409: moh_delete_ccb: called dstadr 0.0.0.0, callid 0 Any ideas? May be the QoS?? Thank you De : Ashar Siddiqui [siddas...@gmail.com] Date d'envoi : vendredi 25 juin 2010 18:51 À : naoufal kerboute Cc : ccie_voice@onlinestudylist.com Objet : Re: [OSL | CCIE_Voice] Multicast Moh from Flash of BR1 First of all, why are you playing? You should be labbing properly as voice field is not a playground.. Joke aside... ;) What is your ‘show ccm-manager music-on-hold ? If no MoH streams are shown by this command then CCM has failed to provide the gateway with MoH Also keep in mind Tone on hold means there is a CM misconfiguration. Silence means the RTP is not getting to the router. Your case is CCM configuration issue. Check configuration again properly. Check at Pub CCM the MOH server - region/Device pool selected properly, reset it and then login to subscriber..go to the same place and reset it from there as well. Also restart Media Voice app. The output you posted means that you have proper telephony-service configuration (max-dn/max-ephone etc)...problem is at cucm side.. You can now continue playing.. Ash naoufal kerboute wrote: Hi, I'm playing with multicast moh over the flash of router, the configuration looks good and I can see in the debug the moh traffic, but on the PSTN phone I heard only bips. below the output of debug ephone moh: Jun 25 18:03:04.729: MoH route If Vlan240 ETHERNET 10.10.201.1 via ARP Jun 25 18:03:04.729: MoH route If Loopback0 46 10.10.110.2 via 10.10.110.2 Jun 25 18:03:09.173: ifs_read flash:music-on-hold.au end of file at 492545 read 3976 = 496521 Jun 25 18:03:09.177: moh tail fill from 24 at 0x4A0E9FF8 length 4024 Jun 25 18:03:10.017: MoH route If Vlan240 ETHERNET 10.10.201.1 via ARP Jun 25 18:03:10.017: MoH route If Loopback0 46 10.10.110.2 via 10.10.110.2 I've created a region who use g711 with other regions, activated the multicast on the source file and the moh server. Any Ideas? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Mobile Voice Acess
Hi, You need only the pin. When you call the MVA number the cucm Check your number with the remote destination profile, if match the mva ask you for the pin. I hope this will help you. Naoufal Envoyé de mon iPhone Le 17 mai 2010 à 21:43, Divin Mathew John divinj...@gmail.com a écrit : Is it possible to configure Mobile Voice access, wihtout the need to enter username and PIN before the user places a call? If yes tell me where do I tweak. Cheers Divin ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCIE Voice Schedule
90 days before the exam date if you Will pay by wire transfer, you CAN schedule and pay by Visa card, it's better Envoyé de mon iPhone Le 17 mai 2010 à 22:10, akash patel akashapa...@yahoo.com a écrit : I am planning to take lab in couple months. I called Cisco Support and they told me that you can't schedule your exam less than 90 days ago. Does anyone know if there is a workaround and schedule the lab whenever the seats are available. Thank you ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] PSTN Router question
Check your partition and css Envoyé de mon iPhone Le 7 févr. 2010 à 00:41, Jefferson Wilson jwil...@annese.com a écrit : Good evening. I am working on Lab 2 from the 4 day instructor led calls this evening (EST) and when I call from the BR2-LON line I get the following from the PSTN router. I am dailling 77353002. I get a fast busy after 7735. I can not call 011442077353003 either. I get a fast busy at the the second 3. Any thoughts. I don’t want to reconfigure the PSTN router. I don’t think we are supposed to. Thank you, Jefferson --More-- Feb 7 01:28:46.731: //-1//DPM/ dpAssociateIncomingPeerCore: Calling Number=02077354765, Called Number=, Voice- Interface=0x486363D8, Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH Feb 7 01:28:46.731: //-1//DPM/ dpAssociateIncomingPeerCore: Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial- peer=20004 Feb 7 01:28:48.291: //-1/F88A661B80A7/DPM/dpMatchPeersCore: Calling Number=, Called Number=7, Peer Info Type=DIALPEER_INFO_SPEECH Feb 7 01:28:48.291: //-1/F88A661B80A7/DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=7 Feb 7 01:28:48.291: //-1/F88A661B80A7/DPM/dpMatchPeersCore: Result=Partial Matches(1) after DP_MATCH_DEST Feb 7 01:28:48.291: //-1/F88A661B80A7/DPM/dpMatchPeersMoreArg: Result=MORE_DIGITS_NEEDED(1) Feb 7 01:28:49.319: //-1/F88A661B80A7/DPM/dpMatchPeersCore: Calling Number=, Called Number=77, Peer Info Type=DIALPEER_INFO_SPEECH Feb 7 01:28:49.319: //-1/F88A661B80A7/DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=77 Feb 7 01:28:49.319: //-1/F88A661B80A7/DPM/dpMatchPeersCore: Result=Partial Matches(1) after DP_MATCH_DEST Feb 7 01:28:49.319: //-1/F88A661B80A7/DPM/dpMatchPeersMoreArg: Result=MORE_DIGITS_NEEDED(1) Feb 7 01:28:51.019: //-1/F88A661B80A7/DPM/dpMatchPeersCore: Calling Number=, Called Number=773, Peer Info Type=DIALPEER_INFO_SPEECH Feb 7 01:28:51.019: //-1/F88A661B80A7/DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=773 Feb 7 01:28:51.019: //-1/F88A661B80A7/DPM/dpMatchPeersCore: No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1) Feb 7 01:28:51.019: //-1/F88A661B80A7/DPM/dpMatchPeersMoreArg: Result=NO_MATCH(-1) ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
