Re: [OSL | CCIE_Voice] I Passed CCIE Voice #27803
Congrats Driss!! Great news. You deserve a merry merry Christmas :) *--* *Dew Swen * On Fri, Dec 24, 2010 at 9:44 AM, Driss BENATTOU driss.benat...@cbi.mawrote: Its a pleasure for me to annonce that i passed CCIE Voice Lab at first attempt in December, 20 at Brussels Location. Thanks to everyone that help me to achieve my goal. Regards Driss BENATTOU CBI Morocco ___ 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] RSVP CAC
Hi Experts, Wish you all and your families MERRY CHRISTMAS and HAPPY HOLIDAYS Below is the scenario. From HQ to BR1 I want to allow only 2 G729 calls. HQ to BR1 call are G729 What I did is : Created MTP on HQ and BR1 routers. dsp profile 2 mtp codec g729r8 codec pass-through maximum sessions soft 2 associate application SCCP and registered them with CUCM. Under serial interfaces: ip rsvp bandwidth 80 On Call manager: HQ - BR1 - Audio ULTD - RSVP setting Mandatory ..vice versa Assigned these locations to DP_HQ and DP_BR1. When I make a call from HQ to BR1 the first call itself says not enough bandwidth My understanding is: Each g729 call need 40K , ip rsvp b/w 80 means 2 calls max session soft 2 - again 2 calls maximum. Why am I getting not enough bandwidth ? T I A Shrini ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] RSVP CAC
Hi Shrini, Do you have the rsvp command in the mtp configuration? HTH Prashant On Fri, Dec 24, 2010 at 2:06 PM, Shrini linuxbos...@gmail.com wrote: Hi Experts, Wish you all and your families MERRY CHRISTMAS and HAPPY HOLIDAYS Below is the scenario. From HQ to BR1 I want to allow only 2 G729 calls. HQ to BR1 call are G729 What I did is : Created MTP on HQ and BR1 routers. dsp profile 2 mtp codec g729r8 codec pass-through maximum sessions soft 2 associate application SCCP and registered them with CUCM. Under serial interfaces: ip rsvp bandwidth 80 On Call manager: HQ - BR1 - Audio ULTD - RSVP setting Mandatory ..vice versa Assigned these locations to DP_HQ and DP_BR1. When I make a call from HQ to BR1 the first call itself says not enough bandwidth My understanding is: Each g729 call need 40K , ip rsvp b/w 80 means 2 calls max session soft 2 - again 2 calls maximum. Why am I getting not enough bandwidth ? T I A Shrini ___ 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] RSVP CAC
Hi Prashant, Yes I forgot to type here but yes it is there on both routers. -S On 12/24/2010 11:53 AM, Prashant Patel wrote: Hi Shrini, Do you have the rsvp command in the mtp configuration? HTH Prashant On Fri, Dec 24, 2010 at 2:06 PM, Shrini linuxbos...@gmail.com mailto:linuxbos...@gmail.com wrote: Hi Experts, Wish you all and your families MERRY CHRISTMAS and HAPPY HOLIDAYS Below is the scenario. From HQ to BR1 I want to allow only 2 G729 calls. HQ to BR1 call are G729 What I did is : Created MTP on HQ and BR1 routers. dsp profile 2 mtp codec g729r8 codec pass-through maximum sessions soft 2 associate application SCCP and registered them with CUCM. Under serial interfaces: ip rsvp bandwidth 80 On Call manager: HQ - BR1 - Audio ULTD - RSVP setting Mandatory ..vice versa Assigned these locations to DP_HQ and DP_BR1. When I make a call from HQ to BR1 the first call itself says not enough bandwidth My understanding is: Each g729 call need 40K , ip rsvp b/w 80 means 2 calls max session soft 2 - again 2 calls maximum. Why am I getting not enough bandwidth ? T I A Shrini ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] RSVP CAC
Hi Shrini, CUCM will always request the worst case scenario in bandwidth first. The easy way to do this is to increase the max bandwidth command and use the show ip rsvp bandwidth command. For example: 1. Increase the ip rsvp bandwidth command to 120 2. Dial a SB phone from HQ, but don't answer 3. Run the show ip rsvp bandwidth command. This will show you the worst case that is being requested by CUCM (in this case 40) 4. Answer the call 5. Run the show ip rsvp bandwidth command, you will see it has dropped back to 24. 6. You then can calculate the value as for 2 calls as 40 +24, or 3 calls 40 + 24 + 24, etc. For your scenario, the value should be ip rsvp bandwidth 64. If this still isn't working then there is something else wrong. Consider these: - If you use MLPP for WAN QoS, you need to move the rsvp command under the Virtual interface. - The ip rsvp bandwidth command should be on both WAN interfaces - Restart the Devices after changing the Location values (although this shouldn't matter, it's still worth a shot) - Run debug ip rsvp and see that RSVP is even being used (although if you can't see output in the show ip rsvp bandwidth as suggested above that will show it as well) - Ensure that the MTPs are registered and in the correct MRGs and MRGLs, reset the MTPs, reset devices in the DPs, etc For help with the debug statements, Matt did a good job of detailing them here and provided a working router configuration - http://matthewberry.info/ciscovoiceguru/377/debug-ip-rsvp-messages/ Hope this helps and Merry Christmas, Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Prashant Patel Sent: Friday, December 24, 2010 11:54 AM To: Shrini Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] RSVP CAC Hi Shrini, Do you have the rsvp command in the mtp configuration? HTH Prashant On Fri, Dec 24, 2010 at 2:06 PM, Shrini linuxbos...@gmail.com wrote: Hi Experts, Wish you all and your families MERRY CHRISTMAS and HAPPY HOLIDAYS Below is the scenario. From HQ to BR1 I want to allow only 2 G729 calls. HQ to BR1 call are G729 What I did is : Created MTP on HQ and BR1 routers. dsp profile 2 mtp codec g729r8 codec pass-through maximum sessions soft 2 associate application SCCP and registered them with CUCM. Under serial interfaces: ip rsvp bandwidth 80 On Call manager: HQ - BR1 - Audio ULTD - RSVP setting Mandatory ..vice versa Assigned these locations to DP_HQ and DP_BR1. When I make a call from HQ to BR1 the first call itself says not enough bandwidth My understanding is: Each g729 call need 40K , ip rsvp b/w 80 means 2 calls max session soft 2 - again 2 calls maximum. Why am I getting not enough bandwidth ? T I A Shrini ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] I Passed CCIE Voice #27803
Awesome. I passed 2 weeks ago but didn't see my post to the group. duy ccie #27737 voice tmobile g2 On Dec 24, 2010 2:21 AM, Dew Swen dew.s...@gmail.com wrote: Congrats Driss!! Great news. You deserve a merry merry Christmas :) *--* *Dew Swen * On Fri, Dec 24, 2010 at 9:44 AM, Driss BENATTOU driss.benat...@cbi.ma wrote: Its a pleasure for me to annonce that i passed CCIE Voice Lab at first attempt in December, 20 at Brussels Location. Thanks to everyone that help me to achieve my goal. Regards Driss BENATTOU CBI Morocco ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 58, Issue 74
I remember doing this Lab and having the same problem, is your BW statement on the actual interface or on the virtual template. I think I moved the BW statement to the virtual template interface and then I was able to make a call without getting not enough BW error display. I was doing so many different things to resolve this That I can't be 100 percent sure though. At the time I also reached out to the list and got the above suggestion. Sent from my iPhone On Dec 24, 2010, at 5:46 PM, ccie_voice-requ...@onlinestudylist.com wrote: Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. RSVP CAC (Shrini) 2. Re: RSVP CAC (Prashant Patel) 3. Re: RSVP CAC (Shrini) 4. Re: RSVP CAC (givemeccievoice2...@gmail.com) -- Message: 1 Date: Fri, 24 Dec 2010 11:06:34 -0800 From: Shrini linuxbos...@gmail.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] RSVP CAC Message-ID: 4d14ef3a.1050...@gmail.com Content-Type: text/plain; charset=iso-8859-1; Format=flowed Hi Experts, Wish you all and your families MERRY CHRISTMAS and HAPPY HOLIDAYS Below is the scenario. From HQ to BR1 I want to allow only 2 G729 calls. HQ to BR1 call are G729 What I did is : Created MTP on HQ and BR1 routers. dsp profile 2 mtp codec g729r8 codec pass-through maximum sessions soft 2 associate application SCCP and registered them with CUCM. Under serial interfaces: ip rsvp bandwidth 80 On Call manager: HQ - BR1 - Audio ULTD - RSVP setting Mandatory ..vice versa Assigned these locations to DP_HQ and DP_BR1. When I make a call from HQ to BR1 the first call itself says not enough bandwidth My understanding is: Each g729 call need 40K , ip rsvp b/w 80 means 2 calls max session soft 2 - again 2 calls maximum. Why am I getting not enough bandwidth ? T I A Shrini -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20101224/c757f726/attachment-0001.html -- Message: 2 Date: Fri, 24 Dec 2010 14:53:34 -0500 From: Prashant Patel prashantpatel...@gmail.com To: Shrini linuxbos...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] RSVP CAC Message-ID: aanlktinfjbsw3vsqer8cnt8-e3i=mrpfjc9rrsnnq...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi Shrini, Do you have the rsvp command in the mtp configuration? HTH Prashant On Fri, Dec 24, 2010 at 2:06 PM, Shrini linuxbos...@gmail.com wrote: Hi Experts, Wish you all and your families MERRY CHRISTMAS and HAPPY HOLIDAYS Below is the scenario. From HQ to BR1 I want to allow only 2 G729 calls. HQ to BR1 call are G729 What I did is : Created MTP on HQ and BR1 routers. dsp profile 2 mtp codec g729r8 codec pass-through maximum sessions soft 2 associate application SCCP and registered them with CUCM. Under serial interfaces: ip rsvp bandwidth 80 On Call manager: HQ - BR1 - Audio ULTD - RSVP setting Mandatory ..vice versa Assigned these locations to DP_HQ and DP_BR1. When I make a call from HQ to BR1 the first call itself says not enough bandwidth My understanding is: Each g729 call need 40K , ip rsvp b/w 80 means 2 calls max session soft 2 - again 2 calls maximum. Why am I getting not enough bandwidth ? T I A Shrini ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20101224/b42f798b/attachment-0001.html -- Message: 3 Date: Fri, 24 Dec 2010 13:39:54 -0800 From: Shrini linuxbos...@gmail.com To: Prashant Patel prashantpatel...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] RSVP CAC Message-ID: 4d15132a.8030...@gmail.com Content-Type: text/plain; charset=iso-8859-1; Format=flowed Hi Prashant, Yes I forgot to type here but yes it is there on both routers. -S On 12/24/2010 11:53 AM, Prashant Patel wrote: Hi Shrini, Do you have the rsvp command in the mtp configuration? HTH Prashant On Fri, Dec 24, 2010 at 2:06 PM, Shrini linuxbos...@gmail.com
[OSL | CCIE_Voice] MOH from three sources
Greetings: I was testing MOH, here is the scenario. HQ (h323) and BR1(mgcp) phones are registered with CUCM. On CUCM I have two audio source files for MOH. Filea - unicast Fileb - Multicast Music On Hold Servers : MOH_SUB - Multicast MOH MOH_PUB - Unicast MOH HQ_MRGL - Have MOH_PUB and BR1_MRGL have MOH_SUB and both are assigned to respective device pools. Problem : Whether I put the call on hold from HQ / BR1 phone I am listening only Filea music playing. I am unable to listen Fileb ( MOH_SUB - Multicast MOH). I tried IP media services shutting down one at a time on PUB and SUB. If I shut SUB not able to listen MOH on BR1 phones and vice versa. Also I have moh on router flash and able to hear when BR1 is in SRST mode not when registered with CUCM, is there any way to play router flash music on CUCM registered phones ? Sorry for so many questions .. T I A Shrini ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] IP Blue Phone with Windows 7
Hello Everyone.. I just wanna check one with with people in this forum..Does IP Blue soft phone works fine wwith Windows 7 operating system or its just me having troubles.. I have tried installing uninstalling and everything.. It doesnt work some time and the error i get is Set the primary call manager IP Out of 10 it works 1 time without me doing anything... Is this a known problem ?? Has anyone exprienced this before ? Thanks Have a great evening.. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
