Re: [OSL | CCIE_Voice] (no subject)
I already done that. Did you do it from user phone with mailbox ? It only works with users with no mailbox. Did anyone answer task 4.4 in lab 7 volume 2 ? Any one can assist me to solve this task. Clear steps will be more accurate. Please see my problem below: Connection between cucm and unity connection is sip trunk. All CSSs of trunk contain partitions of phones. The issue is not with transferring. Users with no mailbox can be transferred to any number they dial during opening greeting, so problem is not with transferring. The problem is with users who have mailboxes. When I press the message button and login, I can't dial any number during the greeting. It says invalid entry. It only allows the predefined options of the greeting to choose from (eg. 1 for new messages, 2 to send messages,etc). Thanks in advance Regards, Waleed Date: Thu, 28 Oct 2010 18:43:09 +0800 From: vcc...@gmail.com To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] (no subject) Under Restriction Table Default System Transfer Uncheck the Blocked checkbox for pattern * It works for me ___ 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] (no subject)
Hi David, Yes that is the question iam trying to answer and I know it asks users with mailbox to do the transfer during opening greeting. I can only do with users who don't have mailbox. I followed the solution guide but didn't work. If you know how, please indicate step by step to follow. I might be missing something. Thanks alot. Regards, Waleed Date: Thu, 28 Oct 2010 18:35:46 -0400 Subject: Re: [OSL | CCIE_Voice] (no subject) From: david.a...@gmail.com To: walid...@hotmail.com CC: vcc...@gmail.com; ccie_voice@onlinestudylist.com Hi Waleed, The question says - Users hqph2 and br1ph2 (mailbox users) press the Messages button and then press # to go get opening greeting. FROM THERE they should be able to dial extensions which do not have a mailbox on unity ie 5001,1001 etc. Is that what you are trying to do ? Its not for users who dont have a mailbox I have done this and it works everytime. Thanks, DA 2010/10/28 Waleed Elhadidy walid...@hotmail.com I already done that. Did you do it from user phone with mailbox ? It only works with users with no mailbox. Did anyone answer task 4.4 in lab 7 volume 2 ? Any one can assist me to solve this task. Clear steps will be more accurate. Please see my problem below: Connection between cucm and unity connection is sip trunk. All CSSs of trunk contain partitions of phones. The issue is not with transferring. Users with no mailbox can be transferred to any number they dial during opening greeting, so problem is not with transferring. The problem is with users who have mailboxes. When I press the message button and login, I can't dial any number during the greeting. It says invalid entry. It only allows the predefined options of the greeting to choose from (eg. 1 for new messages, 2 to send messages,etc). Thanks in advance Regards, Waleed Date: Thu, 28 Oct 2010 18:43:09 +0800 From: vcc...@gmail.com To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] (no subject) Under Restriction Table Default System Transfer Uncheck the Blocked checkbox for pattern * It works for me ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] (no subject)
Hello Everyone, Iam asked to configure unity connection to allow phones with mailbox to press message button and then can be transferred to any number during opening greeting. This can be done by callers with no mailbox but not ones with mailboxes. The solution guide solves this by checking Allow Transfers to Numbers Not Associated with Users or Call Handlers in standard greeting of opening greeting system call handler. I tried this but didn't work. Any ideas ? Regrads, Waleed___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] (no subject)
Hi, Thanks for your tips. I already done the below requirements. What iam facing is when I press the message button and login, I can't dial any number during the greeting. It says invalid entry. It only allows the predefined options of the greeting to choose from. Any other ideas ? Regards, Waleed From: findko...@gmail.com Date: Wed, 27 Oct 2010 12:43:14 +0200 Subject: Re: [OSL | CCIE_Voice] (no subject) To: walid...@hotmail.com CC: ccie_voice@onlinestudylist.com Hi, you also need to adjust the default restriction table and configure proper CSS for the voicemail ports / SIP trunk (+contact line) regards kobel 2010/10/27 Waleed Elhadidy walid...@hotmail.com Hello Everyone, Iam asked to configure unity connection to allow phones with mailbox to press message button and then can be transferred to any number during opening greeting. This can be done by callers with no mailbox but not ones with mailboxes. The solution guide solves this by checking Allow Transfers to Numbers Not Associated with Users or Call Handlers in standard greeting of opening greeting system call handler. I tried this but didn't work. Any ideas ? Regrads, Waleed ___ 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] (no subject)
Hi Prashant, Connection between cucm and unity connection is sip trunk. All CSSs of trunk contain partitions of phones. The issue is not with transferring. Users with no mailbox can be transferred to any number they dial during opening greeting, so problem is not with transferring. The problem is with users who have mailboxes. When I press the message button and login, I can't dial any number during the greeting. It says invalid entry. It only allows the predefined options of the greeting to choose from (eg. 1 for new messages, 2 to send messages,etc). Any other ideas ? Regards, Waleed Date: Wed, 27 Oct 2010 08:26:17 -0400 Subject: Re: [OSL | CCIE_Voice] (no subject) From: prashantpatel...@gmail.com To: walid...@hotmail.com CC: findko...@gmail.com; ccie_voice@onlinestudylist.com Hi Waleed, What is the pt and css you are using for the VM ports. They should be in none partition if you are doing any 4-digit internal number translations. HTH Prashant 2010/10/27 Waleed Elhadidy walid...@hotmail.com Hi, Thanks for your tips. I already done the below requirements. What iam facing is when I press the message button and login, I can't dial any number during the greeting. It says invalid entry. It only allows the predefined options of the greeting to choose from. Any other ideas ? Regards, Waleed From: findko...@gmail.com Date: Wed, 27 Oct 2010 12:43:14 +0200 Subject: Re: [OSL | CCIE_Voice] (no subject) To: walid...@hotmail.com CC: ccie_voice@onlinestudylist.com Hi, you also need to adjust the default restriction table and configure proper CSS for the voicemail ports / SIP trunk (+contact line) regards kobel 2010/10/27 Waleed Elhadidy walid...@hotmail.com Hello Everyone, Iam asked to configure unity connection to allow phones with mailbox to press message button and then can be transferred to any number during opening greeting. This can be done by callers with no mailbox but not ones with mailboxes. The solution guide solves this by checking Allow Transfers to Numbers Not Associated with Users or Call Handlers in standard greeting of opening greeting system call handler. I tried this but didn't work. Any ideas ? Regrads, Waleed ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] (no subject)
I skipped pin when logging and tried pressing # then extension, but still didn't work. The phone menu of the user is what Iam transferred to after login, not opening greeting system handler. This means Iam only allowed to select options in phone menu --Touchtone Conversation--Classic Conversation. Regards, Waleed From: clmar...@bryantx.gov To: walid...@hotmail.com CC: ccie_voice@onlinestudylist.com Date: Wed, 27 Oct 2010 08:33:30 -0500 Subject: RE: [OSL | CCIE_Voice] (no subject) Waleed, Do not enter your password when prompted, just press # that should transfer you to the main greeting then you can dial a number. Chris From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Waleed Elhadidy Sent: Wednesday, October 27, 2010 8:28 AM To: prashantpatel...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] (no subject) Hi Prashant, Connection between cucm and unity connection is sip trunk. All CSSs of trunk contain partitions of phones. The issue is not with transferring. Users with no mailbox can be transferred to any number they dial during opening greeting, so problem is not with transferring. The problem is with users who have mailboxes. When I press the message button and login, I can't dial any number during the greeting. It says invalid entry. It only allows the predefined options of the greeting to choose from (eg. 1 for new messages, 2 to send messages,etc). Any other ideas ? Regards, Waleed Date: Wed, 27 Oct 2010 08:26:17 -0400 Subject: Re: [OSL | CCIE_Voice] (no subject) From: prashantpatel...@gmail.com To: walid...@hotmail.com CC: findko...@gmail.com; ccie_voice@onlinestudylist.com Hi Waleed, What is the pt and css you are using for the VM ports. They should be in none partition if you are doing any 4-digit internal number translations. HTH Prashant 2010/10/27 Waleed Elhadidy walid...@hotmail.com Hi, Thanks for your tips. I already done the below requirements. What iam facing is when I press the message button and login, I can't dial any number during the greeting. It says invalid entry. It only allows the predefined options of the greeting to choose from. Any other ideas ? Regards, Waleed From: findko...@gmail.com Date: Wed, 27 Oct 2010 12:43:14 +0200 Subject: Re: [OSL | CCIE_Voice] (no subject) To: walid...@hotmail.com CC: ccie_voice@onlinestudylist.com Hi, you also need to adjust the default restriction table and configure proper CSS for the voicemail ports / SIP trunk (+contact line) regards kobel 2010/10/27 Waleed Elhadidy walid...@hotmail.com Hello Everyone, Iam asked to configure unity connection to allow phones with mailbox to press message button and then can be transferred to any number during opening greeting. This can be done by callers with no mailbox but not ones with mailboxes. The solution guide solves this by checking Allow Transfers to Numbers Not Associated with Users or Call Handlers in standard greeting of opening greeting system call handler. I tried this but didn't work. Any ideas ? Regrads, Waleed ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] (no subject)
I just want to add something else. The only way it works is when users with no mailbox dial UC because they are transferred to opening greeting system handler, unlike users with mailboxes are transferred to user phone menu. Regards, Waleed From: walid...@hotmail.com To: clmar...@bryantx.gov CC: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] (no subject) Date: Wed, 27 Oct 2010 16:05:30 +0200 I skipped pin when logging and tried pressing # then extension, but still didn't work. The phone menu of the user is what Iam transferred to after login, not opening greeting system handler. This means Iam only allowed to select options in phone menu --Touchtone Conversation--Classic Conversation. Regards, Waleed From: clmar...@bryantx.gov To: walid...@hotmail.com CC: ccie_voice@onlinestudylist.com Date: Wed, 27 Oct 2010 08:33:30 -0500 Subject: RE: [OSL | CCIE_Voice] (no subject) Waleed, Do not enter your password when prompted, just press # that should transfer you to the main greeting then you can dial a number. Chris From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Waleed Elhadidy Sent: Wednesday, October 27, 2010 8:28 AM To: prashantpatel...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] (no subject) Hi Prashant, Connection between cucm and unity connection is sip trunk. All CSSs of trunk contain partitions of phones. The issue is not with transferring. Users with no mailbox can be transferred to any number they dial during opening greeting, so problem is not with transferring. The problem is with users who have mailboxes. When I press the message button and login, I can't dial any number during the greeting. It says invalid entry. It only allows the predefined options of the greeting to choose from (eg. 1 for new messages, 2 to send messages,etc). Any other ideas ? Regards, Waleed Date: Wed, 27 Oct 2010 08:26:17 -0400 Subject: Re: [OSL | CCIE_Voice] (no subject) From: prashantpatel...@gmail.com To: walid...@hotmail.com CC: findko...@gmail.com; ccie_voice@onlinestudylist.com Hi Waleed, What is the pt and css you are using for the VM ports. They should be in none partition if you are doing any 4-digit internal number translations. HTH Prashant 2010/10/27 Waleed Elhadidy walid...@hotmail.com Hi, Thanks for your tips. I already done the below requirements. What iam facing is when I press the message button and login, I can't dial any number during the greeting. It says invalid entry. It only allows the predefined options of the greeting to choose from. Any other ideas ? Regards, Waleed From: findko...@gmail.com Date: Wed, 27 Oct 2010 12:43:14 +0200 Subject: Re: [OSL | CCIE_Voice] (no subject) To: walid...@hotmail.com CC: ccie_voice@onlinestudylist.com Hi, you also need to adjust the default restriction table and configure proper CSS for the voicemail ports / SIP trunk (+contact line) regards kobel 2010/10/27 Waleed Elhadidy walid...@hotmail.com Hello Everyone, Iam asked to configure unity connection to allow phones with mailbox to press message button and then can be transferred to any number during opening greeting. This can be done by callers with no mailbox but not ones with mailboxes. The solution guide solves this by checking Allow Transfers to Numbers Not Associated with Users or Call Handlers in standard greeting of opening greeting system call handler. I tried this but didn't work. Any ideas ? Regrads, Waleed ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Gatekeeper Zone Prefixes
Hello Everyone, I want to solve task 4.2 in lab 6 volume 2. Here is my configuration on HQ-RTR: gatekeeper zone local UCM proctorlabs.com 10.10.110.1 zone prefix UCM 1* gw-priority 10 gw-trunk_2 zone prefix UCM 1* gw-priority 9 gw-trunk_1 zone prefix UCM 1* gw-priority 0 UCME zone prefix UCM 3* gw-priority 10 UCME zone prefix UCM 3* gw-priority 0 gw-trunk_2 gw-trunk_1 zone prefix UCM 5* gw-priority 10 gw-trunk_2 zone prefix UCM 5* gw-priority 9 gw-trunk_1 zone prefix UCM 5* gw-priority 0 UCME no shutdown I wonder why sh gatekeeper gw-type-prefix shows this: GATEWAY TYPE PREFIX TABLE = Prefix: 1#* Zone UCM master gateway list: 10.10.110.3:1720 UCME 10.10.210.11:36373 gk-trunk_2 10.10.210.10:42164 gk-trunk_1 Zone UCM prefix 5* priority gateway list(s): Priority 5: 10.10.210.11:36373 gk-trunk_2 10.10.210.10:42164 gk-trunk_1 Zone UCM prefix 1* priority gateway list(s): Priority 5: 10.10.210.11:36373 gk-trunk_2 10.10.210.10:42164 gk-trunk_1 Zone UCM prefix 3* priority gateway list(s): Priority 10: 10.10.110.3:1720 UCME Priority 5: 10.10.210.11:36373 gk-trunk_2 10.10.210.10:42164 gk-trunk_1 Why is there a mismatch between the configuration and the show command ? Can anyone Please advice ? Regards, Thanks alot Waleed___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Gatekeeper Zone Prefixes
Thanks alot for your help TN. Waleed Date: Sun, 24 Oct 2010 09:17:57 -0500 From: tamnhu...@gmail.com To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Gatekeeper Zone Prefixes Hi Waleed, You need to remove the default priority 5 for the UCM endpoints. Add these two commands in gatekeeper: zone prefix UCM 5*gw-default-priority 0 zone prefix UCM 1*gw-default-priority 0 Regards, TN. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Location CAC
Hi Prashant On site A cac is set to 48k with g729 where the voice mail belongs while site B is set to 32k with g729. You are right about AAR being invoked if inter-region codec is g711, but the codec configured is g729. I could change cac to unlimited but in LAB iam requested to make it 32k. Iam just wondering why AAR is invoked while redirected call should consume only 24k to reach voice mail. Regards, Waleed Date: Wed, 20 Oct 2010 09:31:18 -0400 Subject: Re: [OSL | CCIE_Voice] Location CAC From: prashantpatel...@gmail.com To: walid...@hotmail.com CC: ccie_voice@onlinestudylist.com Hi Waleed, Do you have a Location on the A set as well where the Unity is connected ? If you have A set as 32k and the region codec on A is 711 intra then the call will go as an AAR call. If so make A as Hub-None or unlimited. HTH Prashant 2010/10/20 Waleed Elhadidy walid...@hotmail.com Dears, I just want to understand a point. I configured location cac on cucm between site A and site B to be 32kbps using g729 as codec. When I make a call from pstn to site B phone, it rings until it is redirected to voice mail which is on site A. What happens is that the call is redirected to voice mail via AAR out from site B to pstn to site A. Why does the redirected call invokes AAR while the location cac should allow it to go through WAN from phone at site B to voice mail on site A. The bandwidth required to go through WAN between the 2 regions is 24kbps which is covered by location cac 32kbps. Please advice. Thanks. Waleed ___ 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] Location CAC
Hi, Voice mail is in device pool with region A set to g729 with region B and vice versa. Regards, Waleed From: tih...@gmail.com To: walid...@hotmail.com CC: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Location CAC Date: Wed, 20 Oct 2010 15:35:52 +0200 Hi, Check the codec you used to call voice mail from siteB Tamer, From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Waleed Elhadidy Sent: Wednesday, October 20, 2010 2:33 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Location CAC Dears, I just want to understand a point. I configured location cac on cucm between site A and site B to be 32kbps using g729 as codec. When I make a call from pstn to site B phone, it rings until it is redirected to voice mail which is on site A. What happens is that the call is redirected to voice mail via AAR out from site B to pstn to site A. Why does the redirected call invokes AAR while the location cac should allow it to go through WAN from phone at site B to voice mail on site A. The bandwidth required to go through WAN between the 2 regions is 24kbps which is covered by location cac 32kbps. Please advice. Thanks. Waleed___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Location CAC
Hi Prashant Unfortunately Iam using softphones. If I change location of voice mail in device to hub none it will work normally. In the case of cac set to 32 on site B which is more than needed for g729 call, why is AAR invoked when a call to site B from pstn is redirected to voice mail which is on site A ? Regards, Waleed Date: Wed, 20 Oct 2010 09:58:28 -0400 Subject: Re: [OSL | CCIE_Voice] Location CAC From: prashantpatel...@gmail.com To: walid...@hotmail.com CC: tih...@gmail.com; ccie_voice@onlinestudylist.com Hi Waleed, Even if the requirement is to have 2 g729 calls between A B and A is the Hub/HQ site there should be no CAC on HQ - it should be unlimited. Also when you call Voicemail from HQ whatis the codec used on the phone (pressing ? twice). I hope you are using regular hard phones. HTH Prashant 2010/10/20 Waleed Elhadidy walid...@hotmail.com Hi, Voice mail is in device pool with region A set to g729 with region B and vice versa. Regards, Waleed From: tih...@gmail.com To: walid...@hotmail.com CC: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Location CAC Date: Wed, 20 Oct 2010 15:35:52 +0200 Hi, Check the codec you used to call voice mail from siteB Tamer, From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Waleed Elhadidy Sent: Wednesday, October 20, 2010 2:33 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Location CAC Dears, I just want to understand a point. I configured location cac on cucm between site A and site B to be 32kbps using g729 as codec. When I make a call from pstn to site B phone, it rings until it is redirected to voice mail which is on site A. What happens is that the call is redirected to voice mail via AAR out from site B to pstn to site A. Why does the redirected call invokes AAR while the location cac should allow it to go through WAN from phone at site B to voice mail on site A. The bandwidth required to go through WAN between the 2 regions is 24kbps which is covered by location cac 32kbps. Please advice. Thanks. Waleed ___ 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] Location CAC
By the way if I make a call from site B to site A via WAN, call is established successfully. If I make two calls, only one is allowed which makes sense since cac is set to 32k which allows 1 call (24k). Regards, Waleed From: walid...@hotmail.com To: prashantpatel...@gmail.com CC: tih...@gmail.com; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Location CAC Date: Wed, 20 Oct 2010 16:18:47 +0200 Hi Prashant Unfortunately Iam using softphones. If I change location of voice mail in device to hub none it will work normally. In the case of cac set to 32 on site B which is more than needed for g729 call, why is AAR invoked when a call to site B from pstn is redirected to voice mail which is on site A ? Regards, Waleed Date: Wed, 20 Oct 2010 09:58:28 -0400 Subject: Re: [OSL | CCIE_Voice] Location CAC From: prashantpatel...@gmail.com To: walid...@hotmail.com CC: tih...@gmail.com; ccie_voice@onlinestudylist.com Hi Waleed, Even if the requirement is to have 2 g729 calls between A B and A is the Hub/HQ site there should be no CAC on HQ - it should be unlimited. Also when you call Voicemail from HQ whatis the codec used on the phone (pressing ? twice). I hope you are using regular hard phones. HTH Prashant 2010/10/20 Waleed Elhadidy walid...@hotmail.com Hi, Voice mail is in device pool with region A set to g729 with region B and vice versa. Regards, Waleed From: tih...@gmail.com To: walid...@hotmail.com CC: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Location CAC Date: Wed, 20 Oct 2010 15:35:52 +0200 Hi, Check the codec you used to call voice mail from siteB Tamer, From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Waleed Elhadidy Sent: Wednesday, October 20, 2010 2:33 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Location CAC Dears, I just want to understand a point. I configured location cac on cucm between site A and site B to be 32kbps using g729 as codec. When I make a call from pstn to site B phone, it rings until it is redirected to voice mail which is on site A. What happens is that the call is redirected to voice mail via AAR out from site B to pstn to site A. Why does the redirected call invokes AAR while the location cac should allow it to go through WAN from phone at site B to voice mail on site A. The bandwidth required to go through WAN between the 2 regions is 24kbps which is covered by location cac 32kbps. Please advice. Thanks. Waleed ___ 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] Location CAC
Hi Prashant Thanks for the clarification. I met this situation on vol 2 lab 4 task 4.3. The cac for HQ and BR1 is set to 48k which allows 2 calls. Task 4.3 asks to implement AAR if there is WAN congestion between BR1 and HQ when pstn caller wants to leave a message to BR1 user and BR1 user redirects call to voice mail because of no answer. To verify the AAR feature, it asks to lower cac to 32k and as a result AAR is invoked but don't understand how ? Regards, Waleed Date: Wed, 20 Oct 2010 10:31:12 -0400 Subject: Re: [OSL | CCIE_Voice] Location CAC From: prashantpatel...@gmail.com To: walid...@hotmail.com CC: tih...@gmail.com; ccie_voice@onlinestudylist.com Hi Waleed, I have seen this behavior when I have the HQ CAC set and cisco best practice is to have it unlimited. If it works with unlimited lower the CAC on B to 23. By the way for Locations based CAC 24k is what you need. Call should AAR out when 23. HTH Prashant 2010/10/20 Waleed Elhadidy walid...@hotmail.com By the way if I make a call from site B to site A via WAN, call is established successfully. If I make two calls, only one is allowed which makes sense since cac is set to 32k which allows 1 call (24k). Regards, Waleed From: walid...@hotmail.com To: prashantpatel...@gmail.com CC: tih...@gmail.com; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Location CAC Date: Wed, 20 Oct 2010 16:18:47 +0200 Hi Prashant Unfortunately Iam using softphones. If I change location of voice mail in device to hub none it will work normally. In the case of cac set to 32 on site B which is more than needed for g729 call, why is AAR invoked when a call to site B from pstn is redirected to voice mail which is on site A ? Regards, Waleed Date: Wed, 20 Oct 2010 09:58:28 -0400 Subject: Re: [OSL | CCIE_Voice] Location CAC From: prashantpatel...@gmail.com To: walid...@hotmail.com CC: tih...@gmail.com; ccie_voice@onlinestudylist.com Hi Waleed, Even if the requirement is to have 2 g729 calls between A B and A is the Hub/HQ site there should be no CAC on HQ - it should be unlimited. Also when you call Voicemail from HQ whatis the codec used on the phone (pressing ? twice). I hope you are using regular hard phones. HTH Prashant 2010/10/20 Waleed Elhadidy walid...@hotmail.com Hi, Voice mail is in device pool with region A set to g729 with region B and vice versa. Regards, Waleed From: tih...@gmail.com To: walid...@hotmail.com CC: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Location CAC Date: Wed, 20 Oct 2010 15:35:52 +0200 Hi, Check the codec you used to call voice mail from siteB Tamer, From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Waleed Elhadidy Sent: Wednesday, October 20, 2010 2:33 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Location CAC Dears, I just want to understand a point. I configured location cac on cucm between site A and site B to be 32kbps using g729 as codec. When I make a call from pstn to site B phone, it rings until it is redirected to voice mail which is on site A. What happens is that the call is redirected to voice mail via AAR out from site B to pstn to site A. Why does the redirected call invokes AAR while the location cac should allow it to go through WAN from phone at site B to voice mail on site A. The bandwidth required to go through WAN between the 2 regions is 24kbps which is covered by location cac 32kbps. Please advice. Thanks. Waleed ___ 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] Location CAC
Ok so you mean that there is an error and AAR shouldn't work as 32k will allow redirected call go normally through WAN and that the workbook mean 23k ? right ? Regards, Waleed Date: Wed, 20 Oct 2010 10:49:00 -0400 Subject: Re: [OSL | CCIE_Voice] Location CAC From: prashantpatel...@gmail.com To: walid...@hotmail.com CC: tih...@gmail.com; ccie_voice@onlinestudylist.com It appears to be an error but 23k should do it. Try it out. It has always worked for me. HTH, Prashant 2010/10/20 Waleed Elhadidy walid...@hotmail.com Hi Prashant Thanks for the clarification. I met this situation on vol 2 lab 4 task 4.3. The cac for HQ and BR1 is set to 48k which allows 2 calls. Task 4.3 asks to implement AAR if there is WAN congestion between BR1 and HQ when pstn caller wants to leave a message to BR1 user and BR1 user redirects call to voice mail because of no answer. To verify the AAR feature, it asks to lower cac to 32k and as a result AAR is invoked but don't understand how ? Regards, Waleed Date: Wed, 20 Oct 2010 10:31:12 -0400 Subject: Re: [OSL | CCIE_Voice] Location CAC From: prashantpatel...@gmail.com To: walid...@hotmail.com CC: tih...@gmail.com; ccie_voice@onlinestudylist.com Hi Waleed, I have seen this behavior when I have the HQ CAC set and cisco best practice is to have it unlimited. If it works with unlimited lower the CAC on B to 23. By the way for Locations based CAC 24k is what you need. Call should AAR out when 23. HTH Prashant 2010/10/20 Waleed Elhadidy walid...@hotmail.com By the way if I make a call from site B to site A via WAN, call is established successfully. If I make two calls, only one is allowed which makes sense since cac is set to 32k which allows 1 call (24k). Regards, Waleed From: walid...@hotmail.com To: prashantpatel...@gmail.com CC: tih...@gmail.com; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Location CAC Date: Wed, 20 Oct 2010 16:18:47 +0200 Hi Prashant Unfortunately Iam using softphones. If I change location of voice mail in device to hub none it will work normally. In the case of cac set to 32 on site B which is more than needed for g729 call, why is AAR invoked when a call to site B from pstn is redirected to voice mail which is on site A ? Regards, Waleed Date: Wed, 20 Oct 2010 09:58:28 -0400 Subject: Re: [OSL | CCIE_Voice] Location CAC From: prashantpatel...@gmail.com To: walid...@hotmail.com CC: tih...@gmail.com; ccie_voice@onlinestudylist.com Hi Waleed, Even if the requirement is to have 2 g729 calls between A B and A is the Hub/HQ site there should be no CAC on HQ - it should be unlimited. Also when you call Voicemail from HQ whatis the codec used on the phone (pressing ? twice). I hope you are using regular hard phones. HTH Prashant 2010/10/20 Waleed Elhadidy walid...@hotmail.com Hi, Voice mail is in device pool with region A set to g729 with region B and vice versa. Regards, Waleed From: tih...@gmail.com To: walid...@hotmail.com CC: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Location CAC Date: Wed, 20 Oct 2010 15:35:52 +0200 Hi, Check the codec you used to call voice mail from siteB Tamer, From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Waleed Elhadidy Sent: Wednesday, October 20, 2010 2:33 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Location CAC Dears, I just want to understand a point. I configured location cac on cucm between site A and site B to be 32kbps using g729 as codec. When I make a call from pstn to site B phone, it rings until it is redirected to voice mail which is on site A. What happens is that the call is redirected to voice mail via AAR out from site B to pstn to site A. Why does the redirected call invokes AAR while the location cac should allow it to go through WAN from phone at site B to voice mail on site A. The bandwidth required to go through WAN between the 2 regions is 24kbps which is covered by location cac 32kbps. Please advice. Thanks. Waleed ___ 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] 3 site VGs cannot register MGCP
Thanks. I solved this issue. What I did was add the mgcp configuration first on VGs and then add the configuration of mgcp VGs on cucm. I don't know why this caused the VGs to unregister to cucm, but I removed the mgcp configurations on the VGs and added them once again to get the VGs register to cucm. Date: Mon, 18 Oct 2010 08:48:22 -0700 Subject: Re: [OSL | CCIE_Voice] 3 site VGs cannot register MGCP From: cristobalpri...@gmail.com To: walid...@hotmail.com CC: ccie_voice@onlinestudylist.com what does the show-ccmanager show ? 2010/10/18 Waleed Elhadidy walid...@hotmail.com Hello I couldn't register the 3 VGs of the 3 sites as MGCP on CUCM. I don't know what's the problem, but this is the first time I face it. I did it several times and worked successfully. I don't know what can cause this to happen. here is my HQ-RTR config: controller T1 0/0/0 framing esf linecode b8zs pri-group timeslots 1-3,24 service mgcp ! interface Serial0/0/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn bind-l3 ccm-manager no cdp enable voice-port 0/0/0:23 ! 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 2000 service-type mgcp version 0.1 mgcp dtmf-relay voip codec all mode out-of-band mgcp bind control source-interface FastEthernet0/0.20 mgcp bind media source-interface FastEthernet0/0.20 ! mgcp profile default Please advice. Regards, Waleed ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] 3 site VGs cannot register MGCP
Hello I couldn't register the 3 VGs of the 3 sites as MGCP on CUCM. I don't know what's the problem, but this is the first time I face it. I did it several times and worked successfully. I don't know what can cause this to happen. here is my HQ-RTR config: controller T1 0/0/0 framing esf linecode b8zs pri-group timeslots 1-3,24 service mgcp ! interface Serial0/0/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn bind-l3 ccm-manager no cdp enable voice-port 0/0/0:23 ! 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 2000 service-type mgcp version 0.1 mgcp dtmf-relay voip codec all mode out-of-band mgcp bind control source-interface FastEthernet0/0.20 mgcp bind media source-interface FastEthernet0/0.20 ! mgcp profile default Please advice. Regards, Waleed___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
