Re: [OSL | CCIE_Voice] Vol 2 Q5.1
Got it, Thanks Otto for your response. On Sat, Feb 13, 2010 at 10:30 AM, Otto Sanchez o...@ipexpert.com wrote: Hi, I'd say yes if the negotiated codec between the gw and ucm is g.729r8 (default codec for the dial peer), so make sure the gw is using that codec when talking to ucm or at least it's in the list for the voice class codec used by the dial peer, Thanks, On Wed, Feb 10, 2010 at 2:07 PM, vccie2010 vccie2...@gmail.com wrote: Per the PG solutions the dspfarm profile 1 transcode does not show codec g729r8 it only shows g279ar8 and g729abr8 don't we need codec G729r8 statement here since the traffic coming from UCM across GK will be G729r8 ??? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CCIE Voice Vol 1 Lab 5c - Transcoder
I was working through lab 5c and came across the task where I had to configure a transcoder. I am using a 3725 with AIM-30 - one voice pri with 3 channels - one data T1 I try to create the dspfarm profile and get this erro HQ-RTR(config-dspfarm-profile)#codec ? % Unrecognized command Any idea? -- www.ccietalk.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Adjust 3750 Egress Priority Queue Bandwidth
I wanted some thoughts on how others would handle a request to tweak the amount of bandwidth availble to an egress priority queue on a 3750. So for example a request to allocate 25% of available bandwidth for switchports connected to IP phones on the 3750. I have heard suggestions to handle this in the following manner - this is assuming auto qos voip trust cisco-phone has been run on the port already: interface fa 1/0/2 no priority-queue out srr-queue bandwidth shape 4 0 0 0 srr-queue bandwidth share 0 33 33 33 But I'm struggling to see that this meets the requirement. In this configuration we would be enabling shaping of queue 1 and assigning it 25% of available bandwidth. Then assigning remaining bandwidth equally to the remaining three queues. But this does not appear to be meeting the requirement of assigning the priority queue 25% of the bandwidth. We would be assigning the queue that RTP traffic is placed in by default 25% of total bandwidth but the initial no priority queue out command technically disables a strict priority queue and thus it does not seem to fit the requirement. Thoughts? While I struggle to see the disablement of the priority queue as strictly meeting the requirement - I also find no explicit means to allocate the priority queue a strict amount of bandwidth (I.e. the equal if the ingress queue command - mls qos srr-queue input bandwidth 75 25 that could be used to meet this requirement for default priority ingress queue 1. How about skipping the initial no priority queue-out command but only issuing the shape and share commands as specified above? Wouldn't leaving the priority queue enabled and assigning it a shape value of 25% (1/4) satsify the requirement better? Thanks Scott _ Hotmail: Trusted email with powerful SPAM protection. http://clk.atdmt.com/GBL/go/201469227/direct/01/___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Vol 1 - Lab 5 - Task 5.5 / Setting up TEHO for 212 calls from BR1 through HQ H.323 GW
Vol 1 - Lab 5 - Task 5.5 / Setting up TEHO for 212 calls from BR1 through HQ H.323 GW I can get TEHO to work when dialing a 617 area code number from HQ Phone 2, routing the call over the WAN, out the BR1 MGCP gateway. It works like a charm. It appends the + which seems to come from the 9.1617XXX translation pattern in PT-HQ-PSTN. Problem: I cannot get the + to be sent out when setting up TEHO for 212 area code calls from BR1 through HQ's H.323 GW. All of my settings for the BR1 site are identical to the HQ site. My only guess is that TEHO over WAN and out the BR1 MGCP gateway is using MGCP and not H.323. I can append a + using a dial-peer on the H.323 gateway, but I'm not sure if that is the best way to do it. It seems like Ben was saying that however you produce the end results in the lab is all that matters. What do you guys think? Am I missing something? Digital Footprint: Skype: ciscovoiceguru ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 1 - Lab 5 - Task 5.5 / Setting up TEHO for 212 calls from BR1 through HQ H.323 GW
Hey Matthew, That's the expected behavior since h.323 gateways don't support the + character sending, so if you still want to send that character out to the pstn you should handle it from the router itself (for example voice translation rules), You will find more information in: http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cucm/admin/7_0_1/ccmsys/a03rp.html#wp1166491 Thanks, On Sun, Feb 14, 2010 at 2:38 PM, Berry, Matthew J. mjbe...@krollontrack.com wrote: Vol 1 - Lab 5 - Task 5.5 / Setting up TEHO for 212 calls from BR1 through HQ H.323 GW I can get TEHO to work when dialing a 617 area code number from HQ Phone 2, routing the call over the WAN, out the BR1 MGCP gateway. It works like a charm. It appends the + which seems to come from the 9.1617XXX translation pattern in PT-HQ-PSTN. Problem: I cannot get the + to be sent out when setting up TEHO for 212 area code calls from BR1 through HQ's H.323 GW. All of my settings for the BR1 site are identical to the HQ site. My only guess is that TEHO over WAN and out the BR1 MGCP gateway is using MGCP and not H.323. I can append a + using a dial-peer on the H.323 gateway, but I'm not sure if that is the best way to do it. It seems like Ben was saying that however you produce the end results in the lab is all that matters. What do you guys think? Am I missing something? Digital Footprint: Skype: ciscovoiceguru ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCIE Voice Vol 1 Lab 5c - Transcoder
Hi, I don't think the AIM-VOICE-30 supports transcoding or conferencing but voice termination services only, so in this case you may need to install a NM in your 3725 to move on, Thanks, On Sun, Feb 14, 2010 at 12:07 PM, CCIETalk.com cciet...@gmail.com wrote: I was working through lab 5c and came across the task where I had to configure a transcoder. I am using a 3725 with AIM-30 - one voice pri with 3 channels - one data T1 I try to create the dspfarm profile and get this erro HQ-RTR(config-dspfarm-profile)#codec ? % Unrecognized command Any idea? -- www.ccietalk.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] sending CUCM traces to syslog
Hello, Is there a way to send traces directly from CUCM to syslog? Looking at traces via the CLI is really cumbersome. The syslog agent in enterprise parameters is not it. -- Thanks. tnn314.wordpress.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 1 Lab 5A Task 5.2 - Cannot reach the number display
I hope you tried Reset DP and last resort CCM service too ? On Fri, Feb 12, 2010 at 8:50 AM, Steve Denney (stdenney) stden...@cisco.com wrote: Racking my brains a bit over this one... Trying to place a call from BR1 Ph2 (SCCP IP Blue) to 911 (PSTN phone). Getting a fast busy and this display on the phone: “Cannot reach the number”. Calls from HQ Ph2 (SIP CICP) to 911 work fine – so I know the CUCM route pattern for 911 is working. Also, getting the secondary dial tone when the 9 is dialed from BR1 Ph2 – so the phone is definitely hitting the route pattern. The call just never seems to reach the BR1 MGCP gateway. 911 Route Pattern (None partition) points to rl-local-gw, which points to Standard Local Route Group. Device Pool BR1 is configured to use Local Route Group of rg-br1. Route group rg-br1 has the proper GW selected (S0/SU0/ ds...@br-rtr.proctorlabs.com). BR1 GW is cleanly registered to CUCM, with multiple_frame_established. Have bounced no mgcp / mgcp on BR1 GW, and have reset the phone and route list, multiple times. There are no other route patterns configured which could be conflicting. Any thoughts? thx, sd ___ 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] TEHO - RTP packets seen when call haven't been answered yet?
Hi folks, When testing Vol. 1 Lab 10 question 10.3 (IOS QoS), I noticed that when a TEHO call is made (either from HQ or BR1), I see packet-count in the LLQ increment. When I say call is made, I mean the numbers are dialed, the destination phone is ringing but not picked up/answered yet. The LLQ is only configured to match rtp audio packets; it does not match on rtcp. Where do these RTP audio packets coming from (when the call is not even answered yet)? Thanks, Allen Su ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
