Re: [OSL | CCIE_Voice] Huge Discount for ProctorLabs Vouchers
Hi, When technical problems (interruptions) happen , they have to provide extra rack-slots to everybody affected that's why I think racks get full. BTW I saw CCIE voice volume 3 workbook labs inside proctorlabs rack sessions. Is there any IPX vol 3 wb released? Wyan On Wed, Sep 21, 2011 at 9:25 AM, edgar feliz ejfeli...@yahoo.com wrote: Yeah seems a little strange that the racks are all rented for a period of 2 weeks, and they want us to buy vouchers at this time??? I'll Wait. EJF -- *From:* Jon H jon1...@hotmail.com *To:* Marko Milivojevic mar...@ipexpert.com; Ken Wyan kew...@gmail.com *Cc:* OSL Voice ccie_voice@onlinestudylist.com *Sent:* Tuesday, September 20, 2011 7:58 AM *Subject:* Re: [OSL | CCIE_Voice] Huge Discount for ProctorLabs Vouchers They should have a refund policy. Since we buy allot yet cannot use cause they are so busy, what if you pass. Should be a method to return the unused vouchers. Buy allot now cause they are busy and schedule later is not a good plan in my opinion cause you just end up wasting lots of money on something no one will buy and there is no where to sell them. *From:* Marko Milivojevic mar...@ipexpert.com *Sent:* Monday, September 19, 2011 12:10 AM *To:* Ken Wyan kew...@gmail.com *Cc:* OSL Voice ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] Huge Discount for ProctorLabs Vouchers Racks are pretty busy. However, you don't need to schedule the actual sessions to purchase vouchers. You can get them now and use them later. -- Marko Milivojevic - CCIE #18427 Senior Technical Instructor - IPexpert Free CCIE Training: http://bit.ly/vLecturehttp://bit.ly/vLecture Mailto: mar...@ipexpert.commar...@ipexpert.com Telephone: +1.810.326.1444 Community: http://www.ipexpert.com/communities http://www.ipexpert.com/communities :: Sent from my phone. Apologies for errors and brevity. :: On Sep 18, 2011, at 6:33, Ken Wyan kew...@gmail.com wrote: Marko, Hi IPExpert promotion on vRack sessions is excellent. But from http://www.proctorlabs.com/www.proctorlabs.com , all timeslots are not available for scheduling remote sessions. They seems keeping racks offline some days. This cannot be due to all racks being pre-booked becuse I can see these unavailable slots are not available for all other tracks as well. whatever available slots are available for all tracks. Please look into this update us before this promotion period ends. Wyan On Sun, Sep 18, 2011 at 8:11 AM, Cisco Nut rafayc...@gmail.com rafayc...@gmail.com wrote: Thanks On Sat, Sep 17, 2011 at 6:25 PM, Marko Milivojevic mar...@ipexpert.com mar...@ipexpert.com wrote: Hello everyone, You know we try not to SPAM OSL mailing lists with our marketing material (much :-) ), but I thought you may be interested in the deal we have going on this weekend only. We have 50% discount on ProctorLabs vRack rentals when you purchase them through IPexpert and use the discount code VRACKBOGO at checkout! Here's the link: http://www.ipexpert.com/Cisco/CCIE/vRack-Rental http://www.ipexpert.com/Cisco/CCIE/vRack-Rental Happy studies! -- Marko Milivojevic - CCIE #18427 Senior Technical Instructor - IPexpert FREE CCIE training: http://bit.ly/vLecturehttp://bit.ly/vLecture Mailto: mar...@ipexpert.commar...@ipexpert.com Telephone: +1.810.326.1444 Web: http://www.ipexpert.com/http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.com/www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out http://www.platinumplacement.com/www.PlatinumPlacement.comhttp://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.com/www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out http://www.platinumplacement.com/www.PlatinumPlacement.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 http://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 http://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Problem with CUCM and CUC SIP integration
Hi Brian It is checked for outbound calls. I get the following error when I go through the CUCM sdi trace ,not sure what it means though . 09/20/2011 05:38:02.147 CCM|Forwarding - ERROR processCFToVM - both oCdpnVMPN and cdpnVMPN are NULL - clear the call, callKey= 0x19|CLID::StandAloneClusterNID::10.10.210.11CT::2,100,39,1.251203IP::10.10.201.51DEV::SEPFCFBFBCB5FC5LVL::ErrorMASK::0800 09/20/2011 05:38:02.147 CCM|ConnectionManager - wait_AuDisconnectRequest ERROR:NO ENTRY FOUND IN TABLE,CI(44215327,0),dcType=1,IFCreated(0,0),PID(0-0,0-0),IFHandling(0,0),MCNode(0,0)|CLID::StandAloneClusterNID::10.10.210.11CT::2,100,39,1.251203IP::10.10.201.51DEV::SEPFCFBFBCB5FC5LVL::ErrorMASK::0800 From: Brian Mulgrew [mailto:btmulg...@gmail.com] Sent: 20 September 2011 04:32 PM To: Rynard Coetzee Cc: Geoghegan, Stuart; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Problem with CUCM and CUC SIP integration Under Outbound Calls, check the Redirecting Diversion Header Delivery - Outbound check box On Tue, Sep 20, 2011 at 1:00 PM, Rynard Coetzee rynard.coet...@bytes.co.zamailto:rynard.coet...@bytes.co.za wrote: The null partition will by default be added to any CSS that you create ,you will not actually see it in the list but it is there. In any case my CSS activation policy is set to use the same CSS as the device\line which has access to all the numbers on the system. I have pulled the traces from CUCM will go through them and see if I can pick up anything From: Geoghegan, Stuart [mailto:stuart.geoghe...@ngbailey.co.ukmailto:stuart.geoghe...@ngbailey.co.uk] Sent: 20 September 2011 01:54 PM To: Rynard Coetzee; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: RE: Problem with CUCM and CUC SIP integration But do you actually have a CSS set? or what is the configuration of your CSS activation policy on the line? If you have any other CSS' partitions set elsewhere, you will still need a CSS set to see the null partition From: Rynard Coetzee [mailto:rynard.coet...@bytes.co.zamailto:rynard.coet...@bytes.co.za] Sent: 20 September 2011 12:36 To: Geoghegan, Stuart; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: RE: Problem with CUCM and CUC SIP integration Yes ,Voicemail route pattern is in None partition From: Geoghegan, Stuart [mailto:stuart.geoghe...@ngbailey.co.ukmailto:stuart.geoghe...@ngbailey.co.uk] Sent: 20 September 2011 01:11 PM To: Rynard Coetzee; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: RE: Problem with CUCM and CUC SIP integration HI Rynard Does your line forwarding CSS have visibility of the voice mail route pattern? Kind regards Stuart From: Rynard Coetzee [mailto:rynard.coet...@bytes.co.zamailto:rynard.coet...@bytes.co.za] Sent: 20 September 2011 11:47 To: Geoghegan, Stuart; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: RE: Problem with CUCM and CUC SIP integration Hi Stuart Region setting within DP is set to G711 ,between regions it`s G729 ,but the call fails even on calls from the same region. So HQ phone 1 calls HQ phone 2 ,get fastbusy as soon as call gets forwarded to voicemail ,but the if you dial the VM pilot directly from the phone it works fine. From: Geoghegan, Stuart [mailto:stuart.geoghe...@ngbailey.co.ukmailto:stuart.geoghe...@ngbailey.co.uk] Sent: 20 September 2011 12:44 PM To: Rynard Coetzee; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: RE: Problem with CUCM and CUC SIP integration Hi Rynard, What is your Region setting within your device pool for your SIP trunk? What is its relationship to your BR2 site? Fast busy is usually as codec mis-match Do you have an MRGL on your SIP trunk with a transcoder? Kind regards Stuart 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 Rynard Coetzee Sent: 20 September 2011 08:00 To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Problem with CUCM and CUC SIP integration Hi I am trying to get the SIP integration between CUC and CUCM to work ,but I am stuck at the moment . From my HQ and Branch 1 phones I can dial the UC pilot and have UC answer ,I can also sign into the mailboxes. Problem is when I call from any other phone ,I get a fastbusy as soon as the call gets forwarded to UC ,I have used the Unity Port status monitor but it seems that the call does not get to UC and that it is failing somewhere on the CUCM. Any ideas ? Regards Rynard The information contained in this e-mail (and attachments) is confidential. It must not be read, copied, disclosed, printed, forwarded, relied upon or used by any person other than the intended recipient. Unauthorised use, disclosure or copying is strictly prohibited. If you have received this e-mail in
Re: [OSL | CCIE_Voice] Problem with CUCM and CUC SIP integration
Hi Rynard That's interesting output Cdpn - called party number oCdpn - original called party number I think VMPN would mean voicemail pilot number It's almost as if the device you are forwarding from is not assigned the Voicemail profile Can that specific phone reach the correct unity connection user greeting when you press the messages button? And is there a mask of set on the VM Profile? Regards Stuart From: Rynard Coetzee [mailto:rynard.coet...@bytes.co.za] Sent: 21 September 2011 06:02 To: Brian Mulgrew Cc: Geoghegan, Stuart; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Problem with CUCM and CUC SIP integration Hi Brian It is checked for outbound calls. I get the following error when I go through the CUCM sdi trace ,not sure what it means though . 09/20/2011 05:38:02.147 CCM|Forwarding - ERROR processCFToVM - both oCdpnVMPN and cdpnVMPN are NULL - clear the call, callKey= 0x19|CLID::StandAloneClusterNID::10.10.210.11CT::2,100,39,1.251203IP::10.10.201.51DEV::SEPFCFBFBCB5FC5LVL::ErrorMASK::0800 09/20/2011 05:38:02.147 CCM|ConnectionManager - wait_AuDisconnectRequest ERROR:NO ENTRY FOUND IN TABLE,CI(44215327,0),dcType=1,IFCreated(0,0),PID(0-0,0-0),IFHandling(0,0),MCNode(0,0)|CLID::StandAloneClusterNID::10.10.210.11CT::2,100,39,1.251203IP::10.10.201.51DEV::SEPFCFBFBCB5FC5LVL::ErrorMASK::0800 From: Brian Mulgrew [mailto:btmulg...@gmail.com] Sent: 20 September 2011 04:32 PM To: Rynard Coetzee Cc: Geoghegan, Stuart; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Problem with CUCM and CUC SIP integration Under Outbound Calls, check the Redirecting Diversion Header Delivery - Outbound check box On Tue, Sep 20, 2011 at 1:00 PM, Rynard Coetzee rynard.coet...@bytes.co.zamailto:rynard.coet...@bytes.co.za wrote: The null partition will by default be added to any CSS that you create ,you will not actually see it in the list but it is there. In any case my CSS activation policy is set to use the same CSS as the device\line which has access to all the numbers on the system. I have pulled the traces from CUCM will go through them and see if I can pick up anything From: Geoghegan, Stuart [mailto:stuart.geoghe...@ngbailey.co.ukmailto:stuart.geoghe...@ngbailey.co.uk] Sent: 20 September 2011 01:54 PM To: Rynard Coetzee; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: RE: Problem with CUCM and CUC SIP integration But do you actually have a CSS set? or what is the configuration of your CSS activation policy on the line? If you have any other CSS' partitions set elsewhere, you will still need a CSS set to see the null partition From: Rynard Coetzee [mailto:rynard.coet...@bytes.co.zamailto:rynard.coet...@bytes.co.za] Sent: 20 September 2011 12:36 To: Geoghegan, Stuart; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: RE: Problem with CUCM and CUC SIP integration Yes ,Voicemail route pattern is in None partition From: Geoghegan, Stuart [mailto:stuart.geoghe...@ngbailey.co.ukmailto:stuart.geoghe...@ngbailey.co.uk] Sent: 20 September 2011 01:11 PM To: Rynard Coetzee; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: RE: Problem with CUCM and CUC SIP integration HI Rynard Does your line forwarding CSS have visibility of the voice mail route pattern? Kind regards Stuart From: Rynard Coetzee [mailto:rynard.coet...@bytes.co.zamailto:rynard.coet...@bytes.co.za] Sent: 20 September 2011 11:47 To: Geoghegan, Stuart; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: RE: Problem with CUCM and CUC SIP integration Hi Stuart Region setting within DP is set to G711 ,between regions it`s G729 ,but the call fails even on calls from the same region. So HQ phone 1 calls HQ phone 2 ,get fastbusy as soon as call gets forwarded to voicemail ,but the if you dial the VM pilot directly from the phone it works fine. From: Geoghegan, Stuart [mailto:stuart.geoghe...@ngbailey.co.ukmailto:stuart.geoghe...@ngbailey.co.uk] Sent: 20 September 2011 12:44 PM To: Rynard Coetzee; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: RE: Problem with CUCM and CUC SIP integration Hi Rynard, What is your Region setting within your device pool for your SIP trunk? What is its relationship to your BR2 site? Fast busy is usually as codec mis-match Do you have an MRGL on your SIP trunk with a transcoder? Kind regards Stuart 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 Rynard Coetzee Sent: 20 September 2011 08:00 To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Problem with CUCM and CUC SIP integration Hi I am trying to get the SIP integration between CUC and CUCM to work ,but I am stuck at the moment . From my HQ and Branch 1 phones I
Re: [OSL | CCIE_Voice] Problem with CUCM and CUC SIP integration
Yes the correct VM profile is assigned to the phone ,also the mask has been configured as in the VM profile. When the phone presses the Messages button ,UC answers and I can sign the user into the associated mailbox without any issues ,it is only on forwarding to UC when the problem occurs. From: Geoghegan, Stuart [mailto:stuart.geoghe...@ngbailey.co.uk] Sent: 21 September 2011 10:14 AM To: Rynard Coetzee; Brian Mulgrew Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Problem with CUCM and CUC SIP integration Hi Rynard That's interesting output Cdpn - called party number oCdpn - original called party number I think VMPN would mean voicemail pilot number It's almost as if the device you are forwarding from is not assigned the Voicemail profile Can that specific phone reach the correct unity connection user greeting when you press the messages button? And is there a mask of set on the VM Profile? Regards Stuart From: Rynard Coetzee [mailto:rynard.coet...@bytes.co.za] Sent: 21 September 2011 06:02 To: Brian Mulgrew Cc: Geoghegan, Stuart; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Problem with CUCM and CUC SIP integration Hi Brian It is checked for outbound calls. I get the following error when I go through the CUCM sdi trace ,not sure what it means though . 09/20/2011 05:38:02.147 CCM|Forwarding - ERROR processCFToVM - both oCdpnVMPN and cdpnVMPN are NULL - clear the call, callKey= 0x19|CLID::StandAloneClusterNID::10.10.210.11CT::2,100,39,1.251203IP::10.10.201.51DEV::SEPFCFBFBCB5FC5LVL::ErrorMASK::0800 09/20/2011 05:38:02.147 CCM|ConnectionManager - wait_AuDisconnectRequest ERROR:NO ENTRY FOUND IN TABLE,CI(44215327,0),dcType=1,IFCreated(0,0),PID(0-0,0-0),IFHandling(0,0),MCNode(0,0)|CLID::StandAloneClusterNID::10.10.210.11CT::2,100,39,1.251203IP::10.10.201.51DEV::SEPFCFBFBCB5FC5LVL::ErrorMASK::0800 From: Brian Mulgrew [mailto:btmulg...@gmail.com] Sent: 20 September 2011 04:32 PM To: Rynard Coetzee Cc: Geoghegan, Stuart; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Problem with CUCM and CUC SIP integration Under Outbound Calls, check the Redirecting Diversion Header Delivery - Outbound check box On Tue, Sep 20, 2011 at 1:00 PM, Rynard Coetzee rynard.coet...@bytes.co.zamailto:rynard.coet...@bytes.co.za wrote: The null partition will by default be added to any CSS that you create ,you will not actually see it in the list but it is there. In any case my CSS activation policy is set to use the same CSS as the device\line which has access to all the numbers on the system. I have pulled the traces from CUCM will go through them and see if I can pick up anything From: Geoghegan, Stuart [mailto:stuart.geoghe...@ngbailey.co.ukmailto:stuart.geoghe...@ngbailey.co.uk] Sent: 20 September 2011 01:54 PM To: Rynard Coetzee; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: RE: Problem with CUCM and CUC SIP integration But do you actually have a CSS set? or what is the configuration of your CSS activation policy on the line? If you have any other CSS' partitions set elsewhere, you will still need a CSS set to see the null partition From: Rynard Coetzee [mailto:rynard.coet...@bytes.co.zamailto:rynard.coet...@bytes.co.za] Sent: 20 September 2011 12:36 To: Geoghegan, Stuart; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: RE: Problem with CUCM and CUC SIP integration Yes ,Voicemail route pattern is in None partition From: Geoghegan, Stuart [mailto:stuart.geoghe...@ngbailey.co.ukmailto:stuart.geoghe...@ngbailey.co.uk] Sent: 20 September 2011 01:11 PM To: Rynard Coetzee; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: RE: Problem with CUCM and CUC SIP integration HI Rynard Does your line forwarding CSS have visibility of the voice mail route pattern? Kind regards Stuart From: Rynard Coetzee [mailto:rynard.coet...@bytes.co.zamailto:rynard.coet...@bytes.co.za] Sent: 20 September 2011 11:47 To: Geoghegan, Stuart; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: RE: Problem with CUCM and CUC SIP integration Hi Stuart Region setting within DP is set to G711 ,between regions it`s G729 ,but the call fails even on calls from the same region. So HQ phone 1 calls HQ phone 2 ,get fastbusy as soon as call gets forwarded to voicemail ,but the if you dial the VM pilot directly from the phone it works fine. From: Geoghegan, Stuart [mailto:stuart.geoghe...@ngbailey.co.ukmailto:stuart.geoghe...@ngbailey.co.uk] Sent: 20 September 2011 12:44 PM To: Rynard Coetzee; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: RE: Problem with CUCM and CUC SIP integration Hi Rynard, What is your Region setting within your device pool for your SIP trunk? What is its relationship to your BR2 site? Fast busy is usually as codec mis-match Do you have an MRGL on your SIP trunk with a transcoder? Kind
Re: [OSL | CCIE_Voice] Problem with CUCM and CUC SIP integration
Hi I managed to fix the issue ,it would seem as if my CUCM might be bugged ,I put the VM profile under the Line of the phone and that solved the problem ,my UC VM profile was set to the default profile for the system but it did not work with setting the VM profile under the line to None ,which should then force it to use the default. I might have to re-image my CUCM. Thanks to Stuart for pointing me in the right direction :) From: Geoghegan, Stuart [mailto:stuart.geoghe...@ngbailey.co.uk] Sent: 21 September 2011 10:14 AM To: Rynard Coetzee; Brian Mulgrew Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Problem with CUCM and CUC SIP integration Hi Rynard That's interesting output Cdpn - called party number oCdpn - original called party number I think VMPN would mean voicemail pilot number It's almost as if the device you are forwarding from is not assigned the Voicemail profile Can that specific phone reach the correct unity connection user greeting when you press the messages button? And is there a mask of set on the VM Profile? Regards Stuart From: Rynard Coetzee [mailto:rynard.coet...@bytes.co.za] Sent: 21 September 2011 06:02 To: Brian Mulgrew Cc: Geoghegan, Stuart; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Problem with CUCM and CUC SIP integration Hi Brian It is checked for outbound calls. I get the following error when I go through the CUCM sdi trace ,not sure what it means though . 09/20/2011 05:38:02.147 CCM|Forwarding - ERROR processCFToVM - both oCdpnVMPN and cdpnVMPN are NULL - clear the call, callKey= 0x19|CLID::StandAloneClusterNID::10.10.210.11CT::2,100,39,1.251203IP::10.10.201.51DEV::SEPFCFBFBCB5FC5LVL::ErrorMASK::0800 09/20/2011 05:38:02.147 CCM|ConnectionManager - wait_AuDisconnectRequest ERROR:NO ENTRY FOUND IN TABLE,CI(44215327,0),dcType=1,IFCreated(0,0),PID(0-0,0-0),IFHandling(0,0),MCNode(0,0)|CLID::StandAloneClusterNID::10.10.210.11CT::2,100,39,1.251203IP::10.10.201.51DEV::SEPFCFBFBCB5FC5LVL::ErrorMASK::0800 From: Brian Mulgrew [mailto:btmulg...@gmail.com] Sent: 20 September 2011 04:32 PM To: Rynard Coetzee Cc: Geoghegan, Stuart; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Problem with CUCM and CUC SIP integration Under Outbound Calls, check the Redirecting Diversion Header Delivery - Outbound check box On Tue, Sep 20, 2011 at 1:00 PM, Rynard Coetzee rynard.coet...@bytes.co.zamailto:rynard.coet...@bytes.co.za wrote: The null partition will by default be added to any CSS that you create ,you will not actually see it in the list but it is there. In any case my CSS activation policy is set to use the same CSS as the device\line which has access to all the numbers on the system. I have pulled the traces from CUCM will go through them and see if I can pick up anything From: Geoghegan, Stuart [mailto:stuart.geoghe...@ngbailey.co.ukmailto:stuart.geoghe...@ngbailey.co.uk] Sent: 20 September 2011 01:54 PM To: Rynard Coetzee; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: RE: Problem with CUCM and CUC SIP integration But do you actually have a CSS set? or what is the configuration of your CSS activation policy on the line? If you have any other CSS' partitions set elsewhere, you will still need a CSS set to see the null partition From: Rynard Coetzee [mailto:rynard.coet...@bytes.co.zamailto:rynard.coet...@bytes.co.za] Sent: 20 September 2011 12:36 To: Geoghegan, Stuart; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: RE: Problem with CUCM and CUC SIP integration Yes ,Voicemail route pattern is in None partition From: Geoghegan, Stuart [mailto:stuart.geoghe...@ngbailey.co.ukmailto:stuart.geoghe...@ngbailey.co.uk] Sent: 20 September 2011 01:11 PM To: Rynard Coetzee; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: RE: Problem with CUCM and CUC SIP integration HI Rynard Does your line forwarding CSS have visibility of the voice mail route pattern? Kind regards Stuart From: Rynard Coetzee [mailto:rynard.coet...@bytes.co.zamailto:rynard.coet...@bytes.co.za] Sent: 20 September 2011 11:47 To: Geoghegan, Stuart; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: RE: Problem with CUCM and CUC SIP integration Hi Stuart Region setting within DP is set to G711 ,between regions it`s G729 ,but the call fails even on calls from the same region. So HQ phone 1 calls HQ phone 2 ,get fastbusy as soon as call gets forwarded to voicemail ,but the if you dial the VM pilot directly from the phone it works fine. From: Geoghegan, Stuart [mailto:stuart.geoghe...@ngbailey.co.ukmailto:stuart.geoghe...@ngbailey.co.uk] Sent: 20 September 2011 12:44 PM To: Rynard Coetzee; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: RE: Problem with CUCM and CUC SIP integration Hi Rynard, What is your Region setting within your device pool for your SIP trunk? What is its
Re: [OSL | CCIE_Voice] Problem with CUCM and CUC SIP integration
Hi Rynard Good news, glad to help! Cheers Stuart From: Rynard Coetzee [mailto:rynard.coet...@bytes.co.za] Sent: 21 September 2011 09:50 To: Geoghegan, Stuart; Brian Mulgrew Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Problem with CUCM and CUC SIP integration Hi I managed to fix the issue ,it would seem as if my CUCM might be bugged ,I put the VM profile under the Line of the phone and that solved the problem ,my UC VM profile was set to the default profile for the system but it did not work with setting the VM profile under the line to None ,which should then force it to use the default. I might have to re-image my CUCM. Thanks to Stuart for pointing me in the right direction :) From: Geoghegan, Stuart [mailto:stuart.geoghe...@ngbailey.co.uk] Sent: 21 September 2011 10:14 AM To: Rynard Coetzee; Brian Mulgrew Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Problem with CUCM and CUC SIP integration Hi Rynard That's interesting output Cdpn - called party number oCdpn - original called party number I think VMPN would mean voicemail pilot number It's almost as if the device you are forwarding from is not assigned the Voicemail profile Can that specific phone reach the correct unity connection user greeting when you press the messages button? And is there a mask of set on the VM Profile? Regards Stuart From: Rynard Coetzee [mailto:rynard.coet...@bytes.co.za] Sent: 21 September 2011 06:02 To: Brian Mulgrew Cc: Geoghegan, Stuart; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Problem with CUCM and CUC SIP integration Hi Brian It is checked for outbound calls. I get the following error when I go through the CUCM sdi trace ,not sure what it means though . 09/20/2011 05:38:02.147 CCM|Forwarding - ERROR processCFToVM - both oCdpnVMPN and cdpnVMPN are NULL - clear the call, callKey= 0x19|CLID::StandAloneClusterNID::10.10.210.11CT::2,100,39,1.251203IP::10.10.201.51DEV::SEPFCFBFBCB5FC5LVL::ErrorMASK::0800 09/20/2011 05:38:02.147 CCM|ConnectionManager - wait_AuDisconnectRequest ERROR:NO ENTRY FOUND IN TABLE,CI(44215327,0),dcType=1,IFCreated(0,0),PID(0-0,0-0),IFHandling(0,0),MCNode(0,0)|CLID::StandAloneClusterNID::10.10.210.11CT::2,100,39,1.251203IP::10.10.201.51DEV::SEPFCFBFBCB5FC5LVL::ErrorMASK::0800 From: Brian Mulgrew [mailto:btmulg...@gmail.com] Sent: 20 September 2011 04:32 PM To: Rynard Coetzee Cc: Geoghegan, Stuart; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Problem with CUCM and CUC SIP integration Under Outbound Calls, check the Redirecting Diversion Header Delivery - Outbound check box On Tue, Sep 20, 2011 at 1:00 PM, Rynard Coetzee rynard.coet...@bytes.co.zamailto:rynard.coet...@bytes.co.za wrote: The null partition will by default be added to any CSS that you create ,you will not actually see it in the list but it is there. In any case my CSS activation policy is set to use the same CSS as the device\line which has access to all the numbers on the system. I have pulled the traces from CUCM will go through them and see if I can pick up anything From: Geoghegan, Stuart [mailto:stuart.geoghe...@ngbailey.co.ukmailto:stuart.geoghe...@ngbailey.co.uk] Sent: 20 September 2011 01:54 PM To: Rynard Coetzee; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: RE: Problem with CUCM and CUC SIP integration But do you actually have a CSS set? or what is the configuration of your CSS activation policy on the line? If you have any other CSS' partitions set elsewhere, you will still need a CSS set to see the null partition From: Rynard Coetzee [mailto:rynard.coet...@bytes.co.zamailto:rynard.coet...@bytes.co.za] Sent: 20 September 2011 12:36 To: Geoghegan, Stuart; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: RE: Problem with CUCM and CUC SIP integration Yes ,Voicemail route pattern is in None partition From: Geoghegan, Stuart [mailto:stuart.geoghe...@ngbailey.co.ukmailto:stuart.geoghe...@ngbailey.co.uk] Sent: 20 September 2011 01:11 PM To: Rynard Coetzee; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: RE: Problem with CUCM and CUC SIP integration HI Rynard Does your line forwarding CSS have visibility of the voice mail route pattern? Kind regards Stuart From: Rynard Coetzee [mailto:rynard.coet...@bytes.co.zamailto:rynard.coet...@bytes.co.za] Sent: 20 September 2011 11:47 To: Geoghegan, Stuart; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: RE: Problem with CUCM and CUC SIP integration Hi Stuart Region setting within DP is set to G711 ,between regions it`s G729 ,but the call fails even on calls from the same region. So HQ phone 1 calls HQ phone 2 ,get fastbusy as soon as call gets forwarded to voicemail ,but the if you dial the VM pilot directly from the phone it works fine. From: Geoghegan, Stuart
[OSL | CCIE_Voice] MVA Hairpinning
Hi All, I have an issue with MVA and hairpinning. I have followed IPX proctor guide for LAB 6 question 5.3 I have reached the stage where I can dial in to the MVA access number from the PSTN line (2123942123 - simulated mobile phone number of HQPH2 - 5002) and CUCM recognises the associated Remote destination identity of the Remote destination profile. So I am happy that initially the vxml script is passing the correct digits back to CUCM from the H323 GW. Once I have authenticated with my PIN I press 1 to make a call outbound and the I receive silence and shortly after the GW disconnects the call. It's my understanding that the CSS of the Remote destination profile dictates the partitions and thus route patterns that can be used for MVA. This CSS has visibility of the route pattern that I am trying to dial and as I'm using SLRG I'd expect that that the Route pattern matched points to SLRG and would use the SLRG assigned to the Device Pool of the Remote destination Profile. I have also tried this using a specific CSS for the RDP and specific route pattern pointing to a specific Route List/RG but to no avail. I have checked Call Manager traces which reveal the Remote Destination Information number match and then a H.323 call sent to the HQ Gateway, but no digits appear for what I have attempted to dial back outbound Ccapi inout and h225 asn1 traces on the gateway again just reveal the dialled destination of 5010 which is my MVA access number but then no digits for my dialled number. I have checked the SRND guide for the call flow ( I have changed the numbers to match my config) the Mobile Voice Access user on PSTN phone 408 555-7890 dials the Mobile Voice Access enterprise DID DN 2123945010 (step 1). The call comes into the enterprise PSTN gateway (step 2) and is forwarded to Unified CM for call handling (step 3). Unified CM next routes the inbound call to the H.323 VoiceXML gateway (step 4). The user is then prompted by IVR to enter their numeric user ID, PIN, and then a 1 to make a Mobile Voice Access call, followed by the phone number they wish to reach. Again the user enters 9 1 6745738932 as the number they wish to reach (followed by the # sign). In the meantime, the H.323 VoiceXML gateway collects and forwards the user input to Unified CM and then plays the forwarded IVR prompts to the PSTN gateway and the Mobile Voice Access user. Unified CM in turn receives user input, authenticates the user, and forwards appropriate IVR prompts to the H.323 VoiceXML gateway based on user input (step 5). After receiving the number to be dialed, Unified CM generates a call using the user's Remote Destination Profile (step 6). The outbound call to 9 1 6745738932 is routed through the PSTN gateway (step 7). Finally, the call rings at the PSTN destination phone with number 6745738932 (step 8). So I am failing at step 6, but I believe that my RDP CSS is correct and the Route pattern I am matching is correct (I have verified this by dialling the pattern from a device with the same CSS) I do not see the digits that I am dialing in my traces on CUCM which point sback to the GW/MVA script, however the IVR prompts are sent initially and CUCM authenticate using the PIN, so digits are sent back forth between the CUCM and H323 GW. My GW clears down the call, but this must be due to time-out as it reports normal call clearing HQ-RTR# .Sep 21 12:36:25.106: ISDN Se0/1/0:15 Q931: TX - DISCONNECT pd = 8 callref = 0x824C Cause i = 0x8090 - Normal call clearing .Sep 21 12:36:25.118: ISDN Se0/1/0:15 Q931: RX - RELEASE pd = 8 callref = 0x024C .Sep 21 12:36:25.154: ISDN Se0/1/0:15 Q931: TX - RELEASE_COMP pd = 8 callref = 0x824C Also to note I am following the same as IPexperts where I am using the same GW for MGCP and H323. Any help would be appreciated, thanks, kind regards Stuart Geoghegan The information contained in this e-mail (and attachments) is confidential. It must not be read, copied, disclosed, printed, forwarded, relied upon or used by any person other than the intended recipient. Unauthorised use, disclosure or copying is strictly prohibited. If you have received this e-mail in error immediately contact the sender and postmas...@ngbailey.co.uk then permanently delete it and any attachments. NG Bailey may monitor email where it is considered proportionate to any perceived risk. This email has been sent on behalf of an NG Bailey company, if in doubt ask the sender to clarify which company. The NG Bailey companies include NG Bailey Limited (342778), NG Bailey IT Services Limited (2338401), NG Bailey Facilities Services Limited (5472032), Kedington (Northern Ireland) Limited (NI 31145) and NG Bailey Group Limited (1490238). All of the companies, except Kedington (Northern Ireland) Limited, are registered in England with registered office at Denton Hall Ilkley West Yorkshire LS29 0HH. Kedington (Northern Ireland) Limited is
[OSL | CCIE_Voice] Question on Join Across Lines
Hi Guys Does anyone know what construct JAL uses while bridging 2 calls on different Line Buttons? I ask as I need to plan India specific dial plan which shall restrict bridging of VoIP Calls to Local PSTN Calls. I went through Geolocations but so far am not too convinced with its usability as a well constructed dial plan shall never zero in on 2 IP Endpoints which are not allowed to converse with each other in the first place. Do advise. Thanks Mann ___ 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] MVA Hairpinning
Having the same issue here with MGCP and H323 gateway on the HQ router ... No answer for you yet but I feel your pain ... Sent from my iPad KP On Sep 21, 2011, at 5:39 AM, Geoghegan, Stuart stuart.geoghe...@ngbailey.co.uk wrote: Hi All, I have an issue with MVA and hairpinning. I have followed IPX proctor guide for LAB 6 question 5.3 I have reached the stage where I can dial in to the MVA access number from the PSTN line (2123942123 - simulated mobile phone number of HQPH2 – 5002) and CUCM recognises the associated Remote destination identity of the Remote destination profile. So I am happy that initially the vxml script is passing the correct digits back to CUCM from the H323 GW. Once I have authenticated with my PIN I press 1 to make a call outbound and the I receive silence and shortly after the GW disconnects the call. It’s my understanding that the CSS of the Remote destination profile dictates the partitions and thus route patterns that can be used for MVA. This CSS has visibility of the route pattern that I am trying to dial and as I’m using SLRG I’d expect that that the Route pattern matched points to SLRG and would use the SLRG assigned to the Device Pool of the Remote destination Profile. I have also tried this using a specific CSS for the RDP and specific route pattern pointing to a specific Route List/RG but to no avail. I have checked Call Manager traces which reveal the Remote Destination Information number match and then a H.323 call sent to the HQ Gateway, but no digits appear for what I have attempted to dial back outbound Ccapi inout and h225 asn1 traces on the gateway again just reveal the dialled destination of 5010 which is my MVA access number but then no digits for my dialled number. I have checked the SRND guide for the call flow ( I have changed the numbers to match my config) the Mobile Voice Access user on PSTN phone 408 555-7890 dials the Mobile Voice Access enterprise DID DN 2123945010 (step 1). The call comes into the enterprise PSTN gateway (step 2) and is forwarded to Unified CM for call handling (step 3). Unified CM next routes the inbound call to the H.323 VoiceXML gateway (step 4). The user is then prompted by IVR to enter their numeric user ID, PIN, and then a 1 to make a Mobile Voice Access call, followed by the phone number they wish to reach. Again the user enters 9 1 6745738932 as the number they wish to reach (followed by the # sign). In the meantime, the H.323 VoiceXML gateway collects and forwards the user input to Unified CM and then plays the forwarded IVR prompts to the PSTN gateway and the Mobile Voice Access user. Unified CM in turn receives user input, authenticates the user, and forwards appropriate IVR prompts to the H.323 VoiceXML gateway based on user input (step 5). After receiving the number to be dialed, Unified CM generates a call using the user's Remote Destination Profile (step 6). The outbound call to 9 1 6745738932 is routed through the PSTN gateway (step 7). Finally, the call rings at the PSTN destination phone with number 6745738932 (step 8). So I am failing at step 6, but I believe that my RDP CSS is correct and the Route pattern I am matching is correct (I have verified this by dialling the pattern from a device with the same CSS) I do not see the digits that I am dialing in my traces on CUCM which point sback to the GW/MVA script, however the IVR prompts are sent initially and CUCM authenticate using the PIN, so digits are sent back forth between the CUCM and H323 GW. My GW clears down the call, but this must be due to time-out as it reports normal call clearing HQ-RTR# .Sep 21 12:36:25.106: ISDN Se0/1/0:15 Q931: TX - DISCONNECT pd = 8 callref = 0x824C Cause i = 0x8090 - Normal call clearing .Sep 21 12:36:25.118: ISDN Se0/1/0:15 Q931: RX - RELEASE pd = 8 callref = 0x024C .Sep 21 12:36:25.154: ISDN Se0/1/0:15 Q931: TX - RELEASE_COMP pd = 8 callref = 0x824C Also to note I am following the same as IPexperts where I am using the same GW for MGCP and H323. Any help would be appreciated, thanks, kind regards Stuart Geoghegan The information contained in this e-mail (and attachments) is confidential. It must not be read, copied, disclosed, printed, forwarded, relied upon or used by any person other than the intended recipient. Unauthorised use, disclosure or copying is strictly prohibited. If you have received this e-mail in error immediately contact the sender and postmas...@ngbailey.co.uk then permanently delete it and any attachments. NG Bailey may monitor email where it is considered proportionate to any perceived risk. This email has been sent on behalf of an NG Bailey company, if in
Re: [OSL | CCIE_Voice] MVA Hairpinning
Check the ucm service parameter for mva and see what css is being used to route calls.. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Geoghegan, Stuart Sent: Wednesday, September 21, 2011 8:39 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] MVA Hairpinning Hi All, I have an issue with MVA and hairpinning. I have followed IPX proctor guide for LAB 6 question 5.3 I have reached the stage where I can dial in to the MVA access number from the PSTN line (2123942123 - simulated mobile phone number of HQPH2 - 5002) and CUCM recognises the associated Remote destination identity of the Remote destination profile. So I am happy that initially the vxml script is passing the correct digits back to CUCM from the H323 GW. Once I have authenticated with my PIN I press 1 to make a call outbound and the I receive silence and shortly after the GW disconnects the call. It's my understanding that the CSS of the Remote destination profile dictates the partitions and thus route patterns that can be used for MVA. This CSS has visibility of the route pattern that I am trying to dial and as I'm using SLRG I'd expect that that the Route pattern matched points to SLRG and would use the SLRG assigned to the Device Pool of the Remote destination Profile. I have also tried this using a specific CSS for the RDP and specific route pattern pointing to a specific Route List/RG but to no avail. I have checked Call Manager traces which reveal the Remote Destination Information number match and then a H.323 call sent to the HQ Gateway, but no digits appear for what I have attempted to dial back outbound Ccapi inout and h225 asn1 traces on the gateway again just reveal the dialled destination of 5010 which is my MVA access number but then no digits for my dialled number. I have checked the SRND guide for the call flow ( I have changed the numbers to match my config) the Mobile Voice Access user on PSTN phone 408 555-7890 dials the Mobile Voice Access enterprise DID DN 2123945010 (step 1). The call comes into the enterprise PSTN gateway (step 2) and is forwarded to Unified CM for call handling (step 3). Unified CM next routes the inbound call to the H.323 VoiceXML gateway (step 4). The user is then prompted by IVR to enter their numeric user ID, PIN, and then a 1 to make a Mobile Voice Access call, followed by the phone number they wish to reach. Again the user enters 9 1 6745738932 as the number they wish to reach (followed by the # sign). In the meantime, the H.323 VoiceXML gateway collects and forwards the user input to Unified CM and then plays the forwarded IVR prompts to the PSTN gateway and the Mobile Voice Access user. Unified CM in turn receives user input, authenticates the user, and forwards appropriate IVR prompts to the H.323 VoiceXML gateway based on user input (step 5). After receiving the number to be dialed, Unified CM generates a call using the user's Remote Destination Profile (step 6). The outbound call to 9 1 6745738932 is routed through the PSTN gateway (step 7). Finally, the call rings at the PSTN destination phone with number 6745738932 (step 8). So I am failing at step 6, but I believe that my RDP CSS is correct and the Route pattern I am matching is correct (I have verified this by dialling the pattern from a device with the same CSS) I do not see the digits that I am dialing in my traces on CUCM which point sback to the GW/MVA script, however the IVR prompts are sent initially and CUCM authenticate using the PIN, so digits are sent back forth between the CUCM and H323 GW. My GW clears down the call, but this must be due to time-out as it reports normal call clearing HQ-RTR# .Sep 21 12:36:25.106: ISDN Se0/1/0:15 Q931: TX - DISCONNECT pd = 8 callref = 0x824C Cause i = 0x8090 - Normal call clearing .Sep 21 12:36:25.118: ISDN Se0/1/0:15 Q931: RX - RELEASE pd = 8 callref = 0x024C .Sep 21 12:36:25.154: ISDN Se0/1/0:15 Q931: TX - RELEASE_COMP pd = 8 callref = 0x824C Also to note I am following the same as IPexperts where I am using the same GW for MGCP and H323. Any help would be appreciated, thanks, kind regards Stuart Geoghegan The information contained in this e-mail (and attachments) is confidential. It must not be read, copied, disclosed, printed, forwarded, relied upon or used by any person other than the intended recipient. Unauthorised use, disclosure or copying is strictly prohibited. If you have received this e-mail in error immediately contact the sender and postmas...@ngbailey.co.uk then permanently delete it and any attachments. NG Bailey may monitor email where it is considered proportionate to any perceived risk. This email has been sent on behalf of an NG Bailey company, if in doubt ask the sender to clarify which company. The NG Bailey companies include NG Bailey Limited
[OSL | CCIE_Voice] GATEKEEPER Dial Plan - Routing Question
I am working on a Gatekeeper Lab today and have most if the problems resolved, but one is stumping me and I think it's because I 'm frustrated and need another set of eyes on the problem. So Thanks in advance. Here's a basic rundown on what I've got setup. HQ router/site - MGCP controlled, use UCM trunks to communicate to/from Gatekeeper zone prefix 904 BR1 router/site - CME router H323 controlled zone prefix 617 BR2 router/site - uses UCM via SCCP, everything is h323 zone prefix 3214 Default technology prefix 1# I can call to/from each h323 site to the other h323 site with no issues. I can call from HQ mgcp site to BR1 and BR2 with no problem. When I call from BR1 or BR2 to HQ I can't get the call to go thru, basically inbound GK calls to HQ fail. It appears the calls are prefixed with the 1# technology prefix, which is what I want, but how is the easiest way to strip that off, so I match on the DN of the phone I'm trying to call. I know it sounds easy and I'm sure it is, but for some reason I'm perplexed. Thanks in advance. Stephen Manuel ___ 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] MVA Hairpinning
Hi all, My CSS for my H323 Gateway could see the MVA partition but i did not have the internal partiton - doh! KP, hopefully that solves your problem too Kind Regards Stuart From: DeShon Crayton [dcrayto...@comcast.net] Sent: Wednesday, September 21, 2011 7:03 PM To: Geoghegan, Stuart; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] MVA Hairpinning Check the ucm service parameter for mva and see what css is being used to route calls.. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Geoghegan, Stuart Sent: Wednesday, September 21, 2011 8:39 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] MVA Hairpinning Hi All, I have an issue with MVA and hairpinning. I have followed IPX proctor guide for LAB 6 question 5.3 I have reached the stage where I can dial in to the MVA access number from the PSTN line (2123942123 - simulated mobile phone number of HQPH2 – 5002) and CUCM recognises the associated Remote destination identity of the Remote destination profile. So I am happy that initially the vxml script is passing the correct digits back to CUCM from the H323 GW. Once I have authenticated with my PIN I press 1 to make a call outbound and the I receive silence and shortly after the GW disconnects the call. It’s my understanding that the CSS of the Remote destination profile dictates the partitions and thus route patterns that can be used for MVA. This CSS has visibility of the route pattern that I am trying to dial and as I’m using SLRG I’d expect that that the Route pattern matched points to SLRG and would use the SLRG assigned to the Device Pool of the Remote destination Profile. I have also tried this using a specific CSS for the RDP and specific route pattern pointing to a specific Route List/RG but to no avail. I have checked Call Manager traces which reveal the Remote Destination Information number match and then a H.323 call sent to the HQ Gateway, but no digits appear for what I have attempted to dial back outbound Ccapi inout and h225 asn1 traces on the gateway again just reveal the dialled destination of 5010 which is my MVA access number but then no digits for my dialled number. I have checked the SRND guide for the call flow ( I have changed the numbers to match my config) the Mobile Voice Access user on PSTN phone 408 555-7890 dials the Mobile Voice Access enterprise DID DN 2123945010 (step 1). The call comes into the enterprise PSTN gateway (step 2) and is forwarded to Unified CM for call handling (step 3). Unified CM next routes the inbound call to the H.323 VoiceXML gateway (step 4). The user is then prompted by IVR to enter their numeric user ID, PIN, and then a 1 to make a Mobile Voice Access call, followed by the phone number they wish to reach. Again the user enters 9 1 6745738932 as the number they wish to reach (followed by the # sign). In the meantime, the H.323 VoiceXML gateway collects and forwards the user input to Unified CM and then plays the forwarded IVR prompts to the PSTN gateway and the Mobile Voice Access user. Unified CM in turn receives user input, authenticates the user, and forwards appropriate IVR prompts to the H.323 VoiceXML gateway based on user input (step 5). After receiving the number to be dialed, Unified CM generates a call using the user's Remote Destination Profile (step 6). The outbound call to 9 1 6745738932 is routed through the PSTN gateway (step 7). Finally, the call rings at the PSTN destination phone with number 6745738932 (step 8). So I am failing at step 6, but I believe that my RDP CSS is correct and the Route pattern I am matching is correct (I have verified this by dialling the pattern from a device with the same CSS) I do not see the digits that I am dialing in my traces on CUCM which point sback to the GW/MVA script, however the IVR prompts are sent initially and CUCM authenticate using the PIN, so digits are sent back forth between the CUCM and H323 GW. My GW clears down the call, but this must be due to time-out as it reports normal call clearing HQ-RTR# .Sep 21 12:36:25.106: ISDN Se0/1/0:15 Q931: TX - DISCONNECT pd = 8 callref = 0x824C Cause i = 0x8090 - Normal call clearing .Sep 21 12:36:25.118: ISDN Se0/1/0:15 Q931: RX - RELEASE pd = 8 callref = 0x024C .Sep 21 12:36:25.154: ISDN Se0/1/0:15 Q931: TX - RELEASE_COMP pd = 8 callref = 0x824C Also to note I am following the same as IPexperts where I am using the same GW for MGCP and H323. Any help would be appreciated, thanks, kind regards Stuart Geoghegan The information contained in this e-mail (and attachments) is confidential. It must not be read, copied, disclosed, printed, forwarded, relied upon or used by any person other than the intended recipient. Unauthorised use, disclosure or copying is strictly prohibited. If you have received this e-mail in error immediately contact the
Re: [OSL | CCIE_Voice] MVA Hairpinning
GW clears down the call, but this must be due to time-out as it reports normal call clearing HQ-RTR# .Sep 21 12:36:25.106: ISDN Se0/1/0:15 Q931: TX - DISCONNECT pd = 8 callref = 0x824C Cause i = 0x8090 - Normal call clearing .Sep 21 12:36:25.118: ISDN Se0/1/0:15 Q931: RX - RELEASE pd = 8 callref = 0x024C .Sep 21 12:36:25.154: ISDN Se0/1/0:15 Q931: TX - RELEASE_COMP pd = 8 callref = 0x824C Also to note I am following the same as IPexperts where I am using the same GW for MGCP and H323. Any help would be appreciated, thanks, kind regards Stuart Geoghegan The information contained in this e-mail (and attachments) is confidential. It must not be read, copied, disclosed, printed, forwarded, relied upon or used by any person other than the intended recipient. Unauthorised use, disclosure or copying is strictly prohibited. If you have received this e-mail in error immediately contact the sender and postmas...@ngbailey.co.uk then permanently delete it and any attachments. NG Bailey may monitor email where it is considered proportionate to any perceived risk. This email has been sent on behalf of an NG Bailey company, if in doubt ask the sender to clarify which company. The NG Bailey companies include NG Bailey Limited (342778), NG Bailey IT Services Limited (2338401), NG Bailey Facilities Services Limited (5472032), Kedington (Northern Ireland) Limited (NI 31145) and NG Bailey Group Limited (1490238). All of the companies, except Kedington (Northern Ireland) Limited, are registered in England with registered office at Denton Hall Ilkley West Yorkshire LS29 0HH. Kedington (Northern Ireland) Limited is registered in Northern Ireland and its registered office is c/o Carson McDowell, Murray House, Murray Street, Belfast, BT1 6DN. -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20110921/0b60fbe2/attachment-0001.html -- Message: 2 Date: Wed, 21 Sep 2011 21:00:18 +0530 From: Mann Chaddha mann.chad...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Question on Join Across Lines Message-ID: CAPM-tKZjV5yUQT82BqTCWjCrVRHgic=erzakwb_8_1s-oh-...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi Guys Does anyone know what construct JAL uses while bridging 2 calls on different Line Buttons? I ask as I need to plan India specific dial plan which shall restrict bridging of VoIP Calls to Local PSTN Calls. I went through Geolocations but so far am not too convinced with its usability as a well constructed dial plan shall never zero in on 2 IP Endpoints which are not allowed to converse with each other in the first place. Do advise. Thanks Mann -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20110921/80495452/attachment-0001.html -- ___ CCIE_Voice mailing list CCIE_Voice@onlinestudylist.com http://onlinestudylist.com/mailman/listinfo/ccie_voice End of CCIE_Voice Digest, Vol 67, Issue 119 *** ___ 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 Dial Plan - Routing Question
There are a couple ways to strip the tech prefix for incoming calls into UCM, the simplest is the Significant Digits field on your GK trunk. Set from all to 4 (or whatever works for your dial pattern). If this doesnt work for you, ie TEHO, then use a translation pattern, ie: 1#. or 1#.! using predot to strip it off. HTH Chris On Wed, Sep 21, 2011 at 1:42 PM, Stephen Manuel srman...@bellsouth.netwrote: I am working on a Gatekeeper Lab today and have most if the problems resolved, but one is stumping me and I think it’s because I ‘m frustrated and need another set of eyes on the problem. ** ** So Thanks in advance. ** ** Here’s a basic rundown on what I’ve got setup. ** ** HQ router/site - MGCP controlled, use UCM trunks to communicate to/from Gatekeeper zone prefix 904 BR1 router/site - CME router H323 controlled zone prefix 617 BR2 router/site – uses UCM via SCCP, everything is h323 zone prefix 3214** ** Default technology prefix 1# ** ** I can call to/from each h323 site to the other h323 site with no issues. I can call from HQ mgcp site to BR1 and BR2 with no problem. ** ** When I call from BR1 or BR2 to HQ I can’t get the call to go thru, basically inbound GK calls to HQ fail. It appears the calls are prefixed with the 1# technology prefix, which is what I want, but how is the easiest way to strip that off, so I match on the DN of the phone I’m trying to call. ** ** I know it sounds easy and I’m sure it is, but for some reason I’m perplexed. ** ** Thanks in advance. ** ** Stephen Manuel ** ** ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] MVA Hairpinning
Yeah what was pointed out earlier there is a UCM Service Parameter under mobility called Inbound Calling Search Space for Remote Destination by default it is set to use the gateways incoming calling search space, this can be changed to device+line allowing better control over whom they can call. This is one of those things I tend to change anytime I see MVA to be implemented. HTH, Chris On Wed, Sep 21, 2011 at 1:53 PM, Geoghegan, Stuart stuart.geoghe...@ngbailey.co.uk wrote: Hi all, My CSS for my H323 Gateway could see the MVA partition but i did not have the internal partiton - doh! KP, hopefully that solves your problem too Kind Regards Stuart -- *From:* DeShon Crayton [dcrayto...@comcast.net] *Sent:* Wednesday, September 21, 2011 7:03 PM *To:* Geoghegan, Stuart; ccie_voice@onlinestudylist.com *Subject:* RE: [OSL | CCIE_Voice] MVA Hairpinning Check the ucm service parameter for mva and see what css is being used to route calls.. *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Geoghegan, Stuart *Sent:* Wednesday, September 21, 2011 8:39 AM *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] MVA Hairpinning Hi All, I have an issue with MVA and hairpinning. I have followed IPX proctor guide for LAB 6 question 5.3 I have reached the stage where I can dial in to the MVA access number from the PSTN line (2123942123 - simulated mobile phone number of HQPH2 – 5002) and CUCM recognises the associated Remote destination identity of the Remote destination profile. So I am happy that initially the vxml script is passing the correct digits back to CUCM from the H323 GW. Once I have authenticated with my PIN I press 1 to make a call outbound and the I receive silence and shortly after the GW disconnects the call. It’s my understanding that the CSS of the Remote destination profile dictates the partitions and thus route patterns that can be used for MVA. This CSS has visibility of the route pattern that I am trying to dial and as I’m using SLRG I’d expect that that the Route pattern matched points to SLRG and would use the SLRG assigned to the Device Pool of the Remote destination Profile. I have also tried this using a specific CSS for the RDP and specific route pattern pointing to a specific Route List/RG but to no avail. I have checked Call Manager traces which reveal the Remote Destination Information number match and then a H.323 call sent to the HQ Gateway, but no digits appear for what I have attempted to dial back outbound Ccapi inout and h225 asn1 traces on the gateway again just reveal the dialled destination of 5010 which is my MVA access number but then no digits for my dialled number. I have checked the SRND guide for the call flow ( I have changed the numbers to match my config) *the Mobile Voice Access user on PSTN phone 408 555-7890 dials the Mobile Voice Access enterprise DID DN 2123945010 (step 1). * ** *The call comes into the enterprise PSTN gateway (step 2) * ** *and is forwarded to Unified CM for call handling (step 3).* ** * Unified CM next routes the inbound call to the H.323 VoiceXML gateway (step 4). * ** *The user is then prompted by IVR to enter their numeric user ID, PIN, and then a 1 to make a Mobile Voice Access call, followed by the phone number they wish to reach. Again the user enters 9 1 6745738932 as the number they wish to reach (followed by the # sign).** **In the meantime, the H.323 VoiceXML gateway collects and forwards the user input to Unified CM and then plays the forwarded IVR prompts to the PSTN gateway and the Mobile Voice Access user. Unified CM in turn receives user input, authenticates the user, and forwards appropriate IVR prompts to the H.323 VoiceXML gateway based on user input (step 5). * ** *After receiving the number to be dialed, Unified CM generates a call using the user's Remote Destination Profile (step 6). * ** *The outbound call to 9 1 6745738932 is routed through the PSTN gateway (step 7). * ** *Finally, the call rings at the PSTN destination phone with number 6745738932 (step 8).* So I am failing at step 6, but I believe that my RDP CSS is correct and the Route pattern I am matching is correct (I have verified this by dialling the pattern from a device with the same CSS) I do not see the digits that I am dialing in my traces on CUCM which point sback to the GW/MVA script, however the IVR prompts are sent initially and CUCM authenticate using the PIN, so digits are sent back forth between the CUCM and H323 GW. My GW clears down the call, but this must be due to time-out as it reports normal call clearing ** *HQ-RTR#* *.Sep 21 12:36:25.106: ISDN Se0/1/0:15 Q931: TX - DISCONNECT pd = 8 callref = 0x824C * *Cause i = 0x8090 - Normal call clearing* *.Sep 21 12:36:25.118:
Re: [OSL | CCIE_Voice] GATEKEEPER Dial Plan - Routing Question
You ae Right it is pretty easy On the CUCM Trunk confg set in the inbound call set the Significant digit to the number that matches your internal dial-paln for example if you use 4 digit extensions set the significat digits to 4 Dude you have to take a rest From: srman...@bellsouth.net To: ccie_voice@onlinestudylist.com Date: Wed, 21 Sep 2011 14:42:10 -0400 Subject: [OSL | CCIE_Voice] GATEKEEPER Dial Plan - Routing Question I am working on a Gatekeeper Lab today and have most if the problems resolved, but one is stumping me and I think it’s because I ‘m frustrated and need another set of eyes on the problem. So Thanks in advance. Here’s a basic rundown on what I’ve got setup. HQ router/site - MGCP controlled, use UCM trunks to communicate to/from Gatekeeper zone prefix 904 BR1 router/site - CME router H323 controlled zone prefix 617 BR2 router/site – uses UCM via SCCP, everything is h323 zone prefix 3214 Default technology prefix 1# I can call to/from each h323 site to the other h323 site with no issues. I can call from HQ mgcp site to BR1 and BR2 with no problem. When I call from BR1 or BR2 to HQ I can’t get the call to go thru, basically inbound GK calls to HQ fail. It appears the calls are prefixed with the 1# technology prefix, which is what I want, but how is the easiest way to strip that off, so I match on the DN of the phone I’m trying to call. I know it sounds easy and I’m sure it is, but for some reason I’m perplexed. Thanks in advance. Stephen Manuel ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] GATEKEEPER Dial Plan - Routing Question
You can strip off the digits with a translation pattern Hth Randall Sent from my iPhone On Sep 21, 2011, at 11:42 AM, Stephen Manuel srman...@bellsouth.net wrote: I am working on a Gatekeeper Lab today and have most if the problems resolved, but one is stumping me and I think it’s because I ‘m frustrated and need another set of eyes on the problem. So Thanks in advance. Here’s a basic rundown on what I’ve got setup. HQ router/site - MGCP controlled, use UCM trunks to communicate to/from Gatekeeper zone prefix 904 BR1 router/site - CME router H323 controlled zone prefix 617 BR2 router/site – uses UCM via SCCP, everything is h323 zone prefix 3214 Default technology prefix 1# I can call to/from each h323 site to the other h323 site with no issues. I can call from HQ mgcp site to BR1 and BR2 with no problem. When I call from BR1 or BR2 to HQ I can’t get the call to go thru, basically inbound GK calls to HQ fail. It appears the calls are prefixed with the 1# technology prefix, which is what I want, but how is the easiest way to strip that off, so I match on the DN of the phone I’m trying to call. I know it sounds easy and I’m sure it is, but for some reason I’m perplexed. Thanks in advance. Stephen Manuel ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] GATEKEEPER Dial Plan - Routing Question
First, let me say, I do need a rest, it's hard to work on a lab and do real paying work at the same time. Second, thanks to everyone who has replied. Here's what was throwing me. I have 2 sites built in UCM, HQ which is a 10-digit site, npa-nxx of 904-607-.. And BR2 which an international site that has variable length dialing. So on UCM based on what site is being called the significant digits would be different. The only way I could think this would work is thru a translation pattern, similar to the way I have it working on my CME site. Thanks, Stephen Manuel 12212 Reedpond Drive East Jacksonville, FL 32223 Cell Phone: 904-607-4805 From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of fresh ccie Sent: Wednesday, September 21, 2011 6:18 PM To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] GATEKEEPER Dial Plan - Routing Question You ae Right it is pretty easy On the CUCM Trunk confg set in the inbound call set the Significant digit to the number that matches your internal dial-paln for example if you use 4 digit extensions set the significat digits to 4 Dude you have to take a rest _ From: srman...@bellsouth.net To: ccie_voice@onlinestudylist.com Date: Wed, 21 Sep 2011 14:42:10 -0400 Subject: [OSL | CCIE_Voice] GATEKEEPER Dial Plan - Routing Question I am working on a Gatekeeper Lab today and have most if the problems resolved, but one is stumping me and I think it's because I 'm frustrated and need another set of eyes on the problem. So Thanks in advance. Here's a basic rundown on what I've got setup. HQ router/site - MGCP controlled, use UCM trunks to communicate to/from Gatekeeper zone prefix 904 BR1 router/site - CME router H323 controlled zone prefix 617 BR2 router/site - uses UCM via SCCP, everything is h323 zone prefix 3214 Default technology prefix 1# I can call to/from each h323 site to the other h323 site with no issues. I can call from HQ mgcp site to BR1 and BR2 with no problem. When I call from BR1 or BR2 to HQ I can't get the call to go thru, basically inbound GK calls to HQ fail. It appears the calls are prefixed with the 1# technology prefix, which is what I want, but how is the easiest way to strip that off, so I match on the DN of the phone I'm trying to call. I know it sounds easy and I'm sure it is, but for some reason I'm perplexed. Thanks in advance. Stephen Manuel ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
