Re: [OSL | CCIE_Voice] [OSL | Automated Alternative Routing]
I use this method to prevent having to do any additional dial peers in H323. If only one site is using AAR MCGP - strip as may digits as you need on the AAR Route Pattern to match the called party requirements of the lab and use external mask to satisfy the calling party requirements H323 - strip and prefix digits on the called number to match an existing dialpeer on the gateway. Do not do any calling number manipulations as that is already taken care of by the dial peer If both sites are using AAR Attach calling called number transformations to the gateway to achieve everything from above From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Justin Carney Sent: Friday, February 01, 2013 11:56 PM To: ie ravindra Cc: CCIE Study Subject: Re: [OSL | CCIE_Voice] [OSL | Automated Alternative Routing] I noticed a typo in that last email (sorry, I clicked send too soon) - in the 3 options for H323 digit manipulation, I said num-exp will be between the inbound voip dial peer and outbound VOIP dial peer...the outbound dial peer is POTS in this case, not voip. (using two voip dial-peers on both inbound and outbound is CUBE, which is not relevant to AAR configuration). -Justin On Fri, Feb 1, 2013 at 11:46 PM, Justin Carney justin.s.car...@gmail.com wrote: Yes, AAR is triggered on CAC reporting out of bandwidth. (Side note - the phone will display Network Congestion. Rerouting and this is a service parameter that can be customized, in case that is part of the question requirement.) You are also correct that both phones must be registered to the same CUCM cluster. I don't understand if your last sentence is a question - for some reason that call fails to call by extension - if there is a CSS/PT issue where phone A can't see the DN of phone B, AAR will not kick in. Under normal conditions phone A must be able to call phone B, then when there is no more bandwidth (per CAC) AAR will reroute via PSTN. If the phone B were in SRST mode and the WAN was down, not congested, this would instead use CFUR to reroute. I'll answer question 2 first. A common way to achieve AAR is to use a separate CSS/PT just for AAR, along with an AAR Group assigned to both lines (you can assign AAR group to phones for other reasons, but you *must* put the AAR group on the line/DN). When AAR is triggered (CAC), the called phone B's external number mask will be the new DNIS which should be in E.164 format already, and the calling phone A's AAR-CSS will be used to lookup a route for that DNIS. Simply put a \+.! route pattern in your AAR-PT that routes to the LRG, and the AAR-CSS should contain this AAR-PT. This gets the call to the gateway. If the gateway is MGCP, you may need to manipulate the plan/type to match what the PSTN expects (You may also/instead need to have a \+1.! pattern in AAR-PT in the event your MGCP router's PRI expect a 10 digit DNIS.) For H323 don't do any digit manipulation here, used the gateway to perform all manipulations. Question 1, dial peers needed. If using the strategy above, you might not need any new dial peers. For the MGCP sites there are no dial peers on the router so you are done after CUCM routes the call to the gateway in the proper format. For the H323 sites that need to route the AAR call, the DNIS will be the E.164 number when the call gets to the inbound voip dial peer. If you have an existing outbound pots dial peer that will match this E164 number there is nothing extra to do, your AAR call should be working. (make sure you have the appropriate number of digits and type/plan sent to the PSTN for both ANI and DNIS). If your existing dial peers do not match, you have a few options: 1. you could use a translation-profile on the inbound voip dial peer to manipulate the DNIS into something that matches an existing outbound POTS dial peer 1. for example if your DNIS is +1 408 555 1234 tel:%2B1%20408%20555%201234 , you would change the +1 to 91 and you would match the existing long distance outbound dial peer 2. you could add a new outbound dial peer that will match this DNIS (optionally putting a translation-profile on this dial peer if you need a specific plan/type) 1. for example if your DNIS is +1 408 555 1234 tel:%2B1%20408%20555%201234 , you can copy your existing long distance dial peer (9+11 digits) and just remove the 9 (leaving 11 digits) 3. A third option (I would recommend you do NOT use this option) would be to use number expansion to manipulate the DNIS between the inbound voip dial peer match and the outbound voip dial peer match - the reason I don't recommend this is because number expansion ALWAYS takes place between the inbound and outbound dial peers even if you don't want it to. This means if you're not careful it could break something else that was already working correctly. For question 3, TEHO - if you use the method above, your TEHO patterns will
[OSL | CCIE_Voice] Site A to Site C can't leave vm on CUE
Site A to Site C can't leave vm on CUE Site A to Site C configured for g729 Site A routes to Site C using gk in region C PSTN can leave vm on CUE as can other site C phones I am thinking this is a site C transcoding problem CUE only supports g711; right? Site C transcoding detailed below along with ephone: sccp local Vlan102 sccp ccm 10.196.102.1 identifier 1 version 3.1 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register sctrans ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 12 associate application SCCP telephony-service sdspfarm units 1 sdspfarm transcode sessions 20 sdspfarm tag 1 sctrans ephone 3 codec g729r8 dspfarm-assist dial-peer voice 9000 voip mailbox-selection last-redirect-num destination-pattern 3180 (CUE) session protocol sipv2 session target ipv4:10.196.102.2 dtmf-relay sip-notify codec g711ulaw no vad Any assistance is greatly apprecaited. Joe Fearday ___ 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] Site A to Site C can't leave vm on CUE
You need to transfer h323 to sip on SiteC Voice services voip Allow connections h323 to sip Make sure your Site C Gwy register to Site A GK as H323-GWY instead of VOIP-GWY No gateway Gateway You do need a transcoder and it looks like you have it configured mostly correct You do not need ephone 3 Also it is a good practice to use version 6 or higher for sccp ccm configuration or higher Also use bind interface X under sccp ccm group You should also not need mailbox-selection last-redirect-num From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Joe Fearday Sent: Saturday, February 02, 2013 11:28 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Site A to Site C can't leave vm on CUE Site A to Site C can't leave vm on CUE Site A to Site C configured for g729 Site A routes to Site C using gk in region C PSTN can leave vm on CUE as can other site C phones I am thinking this is a site C transcoding problem CUE only supports g711; right? Site C transcoding detailed below along with ephone: sccp local Vlan102 sccp ccm 10.196.102.1 identifier 1 version 3.1 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register sctrans ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 12 associate application SCCP telephony-service sdspfarm units 1 sdspfarm transcode sessions 20 sdspfarm tag 1 sctrans ephone 3 codec g729r8 dspfarm-assist dial-peer voice 9000 voip mailbox-selection last-redirect-num destination-pattern 3180 (CUE) session protocol sipv2 session target ipv4:10.196.102.2 dtmf-relay sip-notify codec g711ulaw no vad Any assistance is greatly apprecaited. Joe Fearday ___ 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] Site A to Site C can't leave vm on CUE
Hi, Check your WAN QoS It could be related to one way RTP compression. Regards Ramcharan Arya On Sat, Feb 2, 2013 at 12:37 PM, Cory Gray corygray22...@hotmail.comwrote: *You need to transfer h323 to sip on SiteC* Voice services voip Allow connections h323 to sip ** ** *Make sure your Site C Gwy register to Site A GK as H323-GWY instead of VOIP-GWY* No gateway Gateway ** ** You do need a transcoder and it looks like you have it configured mostly correct You do not need ephone 3 Also it is a good practice to use version 6 or higher for sccp ccm configuration or higher Also use bind interface X under sccp ccm group You should also not need “mailbox-selection last-redirect-num” ** ** ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Joe Fearday *Sent:* Saturday, February 02, 2013 11:28 AM *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] Site A to Site C can't leave vm on CUE ** ** Site A to Site C can't leave vm on CUE Site A to Site C configured for g729 Site A routes to Site C using gk in region C PSTN can leave vm on CUE as can other site C phones I am thinking this is a site C transcoding problem CUE only supports g711; right? Site C transcoding detailed below along with ephone: sccp local Vlan102 sccp ccm 10.196.102.1 identifier 1 version 3.1 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register sctrans ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 12 associate application SCCP telephony-service sdspfarm units 1 sdspfarm transcode sessions 20 sdspfarm tag 1 sctrans ephone 3 codec g729r8 dspfarm-assist dial-peer voice 9000 voip mailbox-selection last-redirect-num destination-pattern 3180 (CUE) session protocol sipv2 session target ipv4:10.196.102.2 dtmf-relay sip-notify codec g711ulaw no vad Any assistance is greatly apprecaited. Joe Fearday ___ 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
[OSL | CCIE_Voice] Site A to Site C can't leave vm on CUE
Greetings Joe, Could you post or send me a copy of your complete configuration for Site C. Sure sounds like a transcoding issue but who knows. Do calls complete and roll to CUE VM? From Site A? What happens when calls roll to VM get rapid busy??Is transcoder get invoked when you place call through GK? What does the GK dial-peer look like on Site C? --ms Michael Sears Compucom Systems Western Region Senior Consultant Office: +1.720.344.6833 Mobile: +1.303.328.5590 Fax:+1.978.863.0740 [Description: Description: ccnp_voice_sm] inline: image001.jpg___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] [OSL || CCIE VOICE ]
Hi Mates , I have few questions on the real lab just to make up my speed. Thanks for all you guys I believe I have made my hardest part DialPlan two easier. my only clue is now AAR. but thanks for Jsutin and Cory I am evaluate those today. Now I got another problem. I am still following technology based lab approach since I am not too practiced device based aproach yet. But still I am getting 30 Mins to complete DHCP and VLAN section except NTP. will that be a problem in starting minutes in the real lab. Do you guys have any advices on that as well as the whole lab exam what ever the experiences you got in the real lab, Thanks , Ravi. ___ 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] Site A to Site C can't leave vm on CUE
that config looks pretty good, as long as that xcoder is registered. ( sho dspfarm profile ) have you checked under Telephony service , call-foward pattern .T i thought that voip to voip calls into CUE required that command, i struggled with that one night, i think that got it working for me... On Sat, Feb 2, 2013 at 11:27 AM, Joe Fearday feard...@trinity-health.orgwrote: Site A to Site C can't leave vm on CUE Site A to Site C configured for g729 Site A routes to Site C using gk in region C PSTN can leave vm on CUE as can other site C phones I am thinking this is a site C transcoding problem CUE only supports g711; right? Site C transcoding detailed below along with ephone: sccp local Vlan102 sccp ccm 10.196.102.1 identifier 1 version 3.1 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register sctrans ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 12 associate application SCCP telephony-service sdspfarm units 1 sdspfarm transcode sessions 20 sdspfarm tag 1 sctrans ephone 3 codec g729r8 dspfarm-assist dial-peer voice 9000 voip mailbox-selection last-redirect-num destination-pattern 3180 (CUE) session protocol sipv2 session target ipv4:10.196.102.2 dtmf-relay sip-notify codec g711ulaw no vad Any assistance is greatly apprecaited. Joe Fearday ___ 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] [OSL || CCIE VOICE ]
Ravi, You need to speed up. 30 minutes for these sections is too much.. What exactly in these two sections is taking too long for you? usually DHCP and VLANs should not take more than 10 to 15 minutes as far as my practice goes. Practice them over and over to build speed and accuracy. Regards G On Sat, Feb 2, 2013 at 11:11 PM, ie ravindra ieravin...@gmail.com wrote: Hi Mates , I have few questions on the real lab just to make up my speed. Thanks for all you guys I believe I have made my hardest part DialPlan two easier. my only clue is now AAR. but thanks for Jsutin and Cory I am evaluate those today. Now I got another problem. I am still following technology based lab approach since I am not too practiced device based aproach yet. But still I am getting 30 Mins to complete DHCP and VLAN section except NTP. will that be a problem in starting minutes in the real lab. Do you guys have any advices on that as well as the whole lab exam what ever the experiences you got in the real lab, Thanks , Ravi. ___ 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
