Re: [OSL | CCIE_Voice] Lab Exam Speed Strategy
Hi Jamie, Would you please explain this more : ***You have to setup most devices with little or no prior configuration, there are things that cannot change. Know these things and practice them over and over so you do not have to think about them* Thanks in advance On Wed, Mar 27, 2013 at 9:00 PM, Jamie Parr (jamparr) jamp...@cisco.comwrote: First attempt I was very slow – did not use the device based approach, did not finish all tasks. Second I was much faster – using the device based approach, finished with 3 hours to test. Third attempt I finished with more than 3 hours to test and pick up the issues – Passed ** ** My advice: **· **I found the more I practiced the faster I got, practice practice practice **· **Use notepad to write all your device configs first, you can copy and paste large sections of config saving a lot of time **· **Do not be so strict to the device based approach, use it as a base and create your own hybrid **· **You have to setup most devices with little or no prior configuration, there are things that cannot change. Know these things and practice them over and over so you do not have to think about them **· **Persevere, it’s not easy and it sucks most of the time but you will get there ** ** Hope this helps ** ** *Jamie Parr* CCIE #38633 (voice) Engineer - IT jamp...@cisco.com Phone: *+44 20 8824 2641* Mobile: *+44 7590622049* ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Dane Warner *Sent:* 26 March 2013 23:41 *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] Lab Exam Speed Strategy ** ** To All, ** ** I took my second attempt on Monday, March 25 and did not pass. I was hoping for some insight on concrete suggestions to get faster. I didn’t get hung up on any one task, I seemed to keep moving forward and tried to type as fast as I could, using CLI shortcuts, etc. I used the device-based methodology and I feel pretty confident of my technical knowledge. Yet I didn’t even get to many tasks at all, I would have needed another 2-3 hours to complete all tasks. I hear of candidates completing all tasks in 6-7 hours, which means I would need to become twice as fast as my last attempt. It almost sounds insurmountable. Do I need to take typing classes? ** ** Any recommendations that don’t break the NDA would be greatly appreciated. ** ** Regards, ** ** *Dane Warner, CCVP* *Sr. Network Engineer* *Epoch Universal, Inc.* *(909)226-0755* *dwar...@epochuniversal.com * *[image: Epoch_Logo_Smaller_Transparent]* ___ 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 image001.png___ 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] LAB 10.1 VOL1 - RSVP enabled location
Hi Experts, It’s my first email (first impression WOW) I’m recently start preparing for the LAB hopefully I will pass from the first attempt. I’m facing very strange issue with LAB 10.1 VOL1 - RSVP enabled location …the lab is asking to configure and test RSVP between Phone HQ and Phone BR1 using VOIP over WAN and here is the call flow: Phone-HQCUCM ---H323 (HQ-VGW) --H323(BR1-VGW)Phone-BR1 …the VOIP call is working fine and i configured everything correctly (MTP,MRGL….etc) even I have checked DSG for lab but I’m facing the following two issues: 1- 1-I got busy tone once I press any button for supplementary service (Hold, Transfer ….etc) from both sites. (HQ BR1). 2- 2-During VOIP call I cant see any RSVP connection from both VGW. 3- 3-In case I replace BR1-VGW from H323 to MGCP how can I configured VOIP call from Phone-BR1 - MGCP (BR1-VGW) --H323(HQ-VGW)Phone-HQ : I noticed that there is an issue with MGCP (BR1-VGW) once I tried VOIP call. Thanks in advance. T-Joe On Wed, Mar 27, 2013 at 11:27 PM, Tariq Joe tjoe2...@gmail.com wrote: Hi Experts, It’s my first email (first impression WOW) I’m recently start preparing for the LAB hopefully I will pass from the first attempt. I’m facing very strange issue with LAB 10.1 VOL1 - RSVP enabled location …the lab is asking to configure and test RSVP between Phone HQ and Phone BR1 using VOIP over WAN and here is the call flow: Phone-HQCUCM ---H323 (HQ-VGW) --H323(BR1-VGW)Phone-BR1 …the VOIP call is working fine and i configured everything correctly (MTP,MRGL….etc) even I have checked DSG for lab but I’m facing the following two issues: 1- 1-I got busy tone once I press any button for supplementary service (Hold, Transfer ….etc) from both sites. (HQ BR1). 2- 2-During VOIP call I cant see any RSVP connection from both VGW. 3- 3-In case I replace BR1-VGW from H323 to MGCP how can I configured VOIP call from Phone-BR1 - MGCP (BR1-VGW) --H323(HQ-VGW)Phone-HQ : I noticed that there is an issue with MGCP (BR1-VGW) once I tried VOIP call. Thanks in advance. T-Joe ___ 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] Voicemail Ports on SA
I follow the requirements for site A phones but I mask with the number of the VM pilot. So, using ipExpert lab examples: Site A would be +1202555 for the phones Unity Connection VM Pilot number is 2600 I set the external mask on vm ports to +12025552600. -BIll -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 27, 2013, at 9:50 PM, Barrera, Hugo wrote: Hi Guy’s, For the voicemail ports on SA what do you recommend to put for the external mask? Should it match the phones external mask OR should it be only 10 digits because you’re not supposed to send the 1 out of SA? Thoughts would be appreciated? - Hugo ___ 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] B-ACD drop through to loop ephone hunt a second time
Suresh, Make sure you enable the music on hold under the cme. If the moh is not specified the BACD will drop the second time it attempts again. Thanks --- On Thu, 28/3/13, ccie_voice-requ...@onlinestudylist.com ccie_voice-requ...@onlinestudylist.com wrote: From: ccie_voice-requ...@onlinestudylist.com ccie_voice-requ...@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 85, Issue 104 To: ccie_voice@onlinestudylist.com Date: Thursday, 28 March, 2013, 4:08 AM 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. B-ACD drop through to loop ephone hunt a second time (Suresh Bhandari) 2. Re: QOS big question (ikizoo4 kwon) 3. MVA functionality (CCIEing) 4. Re: B-ACD drop through to loop ephone hunt a second time (William Bell) 5. Re: MVA partial match issue (William Bell) -- Message: 1 Date: Thu, 28 Mar 2013 00:56:11 +0545 From: Suresh Bhandari bring...@gmail.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] B-ACD drop through to loop ephone hunt a second time Message-ID: CAExcHf=6g3nivqohbvpnjchxrzccqjd2oyoq2mokoyu+k2f...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Experts! I configured the embedded drop-through script to match the requirement that if, for the first time, both the agents do not pickup the call, it should once more attempt to send the call to the agents. Succeeded for one time only. On the calling phone, I hear the all of our agents ... or so, and goes on hook, never attempts a second time. Somewhere I read to tweak the second-greeting-timer to 35secs or less. Did it, but no avail. can anyone shed light on what should i do to achieve the results? TIA -- Suresh Bhandari -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20130328/d3a0ab13/attachment-0001.html -- Message: 2 Date: Wed, 27 Mar 2013 12:19:30 -0700 From: ikizoo4 kwon ikiz...@hotmail.com To: Suresh Bhandari bring...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com, singh singh8...@in.com, ccie_voice-requ...@onlinestudylist.com ccie_voice-requ...@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] QOS big question Message-ID: bay172-w45ba04dce7f1bb3edccf0cea...@phx.gbl Content-Type: text/plain; charset=iso-8859-1 i am not talking about theory, as you know there is lot of theory around. as you can see i enabled FRF.12 and cRTP , then make 1 g729 call, the bandwidth priority queue has 25K ( not even close to 12) sh policy-map int Serial0/3/0.2 Serial0/3/0.2: DLCI 103 - Service-policy output: AutoQoS-Policy-Trust queue stats for all priority classes: queue limit 64 packets (queue depth/total drops/no-buffer drops) 0/0/0 (pkts output/bytes output) 0/0 Class-map: AutoQoS-VoIP-RTP-Trust (match-any) 15075 packets, 964800 bytes 5 minute offered rate 25000 bps, drop rate 0 bps = 25K Match: ip dscp ef (46) 15075 packets, 964800 bytes 5 minute rate 25000 bps Priority: 70% (537 kbps), burst bytes 13400, b/w exceed drops: 0 compress: header ip rtp UDP/RTP (compression on, Cisco, RTP) Sent: 15075 total, 15074 compressed, == cRTP working 572780 bytes saved, 331720 bytes sent 2.72 efficiency improvement factor 99% hit ratio, five minute miss rate 0 misses/sec, 0 max rate 8000 bps Class-map: AutoQoS-VoIP-Control-Trust (match-any) 343 packets, 20788 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: ip dscp cs3 (24) 343 packets, 20788 bytes 5 minute rate 0 bps Match: ip dscp af31 (26) 0 packets, 0 bytes 5 minute rate 0 bps Queueing queue limit 64 packets (queue depth/total drops/no-buffer drops) 0/0/0 (pkts output/bytes output) 0/0 bandwidth 5% (38 kbps) Class-map: class-default (match-any) 430 packets, 47701 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: any Queueing queue limit 64 packets (queue depth/total drops/no-buffer drops/flowdrops) 0/0/0/0 (pkts output/bytes output) 5/6516 Fair-queue: per-flow queue limit 16 Date: Thu, 28 Mar 2013 00:42:00 +0545 Subject: Re: [OSL
Re: [OSL | CCIE_Voice] B-ACD drop through to loop ephone hunt a second time
You got the beast Ajay! It worked as expected Thanks. On Thu, Mar 28, 2013 at 6:49 PM, Ajay Viswanath ajayviswan...@yahoo.co.inwrote: Suresh, Make sure you enable the music on hold under the cme. If the moh is not specified the BACD will drop the second time it attempts again. Thanks --- On *Thu, 28/3/13, ccie_voice-requ...@onlinestudylist.com ccie_voice-requ...@onlinestudylist.com* wrote: From: ccie_voice-requ...@onlinestudylist.com ccie_voice-requ...@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 85, Issue 104 To: ccie_voice@onlinestudylist.com Date: Thursday, 28 March, 2013, 4:08 AM Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.comhttp://mc/compose?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.comhttp://mc/compose?to=ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.comhttp://mc/compose?to=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. B-ACD drop through to loop ephone hunt a secondtime (Suresh Bhandari) 2. Re: QOS big question (ikizoo4 kwon) 3. MVA functionality (CCIEing) 4. Re: B-ACD drop through to loop ephone hunt asecond time (William Bell) 5. Re: MVA partial match issue (William Bell) -- Message: 1 Date: Thu, 28 Mar 2013 00:56:11 +0545 From: Suresh Bhandari bring...@gmail.comhttp://mc/compose?to=bring...@gmail.com To: ccie_voice@onlinestudylist.comhttp://mc/compose?to=ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.comhttp://mc/compose?to=ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] B-ACD drop through to loop ephone hunt a secondtime Message-ID: CAExcHf=6g3nivqohbvpnjchxrzccqjd2oyoq2mokoyu+k2f...@mail.gmail.comhttp://mc/compose?to=k2f...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Experts! I configured the embedded drop-through script to match the requirement that if, for the first time, both the agents do not pickup the call, it should once more attempt to send the call to the agents. Succeeded for one time only. On the calling phone, I hear the all of our agents ... or so, and goes on hook, never attempts a second time. Somewhere I read to tweak the second-greeting-timer to 35secs or less. Did it, but no avail. can anyone shed light on what should i do to achieve the results? TIA -- Suresh Bhandari -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20130328/d3a0ab13/attachment-0001.html -- Message: 2 Date: Wed, 27 Mar 2013 12:19:30 -0700 From: ikizoo4 kwon ikiz...@hotmail.comhttp://mc/compose?to=ikiz...@hotmail.com To: Suresh Bhandari bring...@gmail.comhttp://mc/compose?to=bring...@gmail.com Cc: ccie_voice@onlinestudylist.comhttp://mc/compose?to=ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.comhttp://mc/compose?to=ccie_voice@onlinestudylist.com , singh singh8...@in.com http://mc/compose?to=singh8...@in.com, ccie_voice-requ...@onlinestudylist.comhttp://mc/compose?to=ccie_voice-requ...@onlinestudylist.com ccie_voice-requ...@onlinestudylist.comhttp://mc/compose?to=ccie_voice-requ...@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] QOS big question Message-ID: bay172-w45ba04dce7f1bb3edccf0cea...@phx.gblhttp://mc/compose?to=bay172-w45ba04dce7f1bb3edccf0cea...@phx.gbl Content-Type: text/plain; charset=iso-8859-1 i am not talking about theory, as you know there is lot of theory around. as you can see i enabled FRF.12 and cRTP , then make 1 g729 call, the bandwidth priority queue has 25K ( not even close to 12) sh policy-map int Serial0/3/0.2 Serial0/3/0.2: DLCI 103 - Service-policy output: AutoQoS-Policy-Trust queue stats for all priority classes: queue limit 64 packets (queue depth/total drops/no-buffer drops) 0/0/0 (pkts output/bytes output) 0/0 Class-map: AutoQoS-VoIP-RTP-Trust (match-any) 15075 packets, 964800 bytes 5 minute offered rate 25000 bps, drop rate 0 bps = 25K Match: ip dscp ef (46) 15075 packets, 964800 bytes5 minute rate 25000 bps Priority: 70% (537 kbps), burst bytes 13400, b/w exceed drops: 0 compress: header ip rtp UDP/RTP (compression on, Cisco, RTP)Sent:15075 total, 15074 compressed, == cRTP working 572780 bytes saved, 331720 bytes sent 2.72
Re: [OSL | CCIE_Voice] Voicemail Ports on SA
Would this only be used if there is a call outbound from Unity and you do not have the service parameter to use original caller id when call routes through unity? Not sure of the exact parameter name , but i think everyone is familiar with that one by now. Thus caller id would be Voicemail/+12025552002 for a call the came from Unity. Even in this case i would think you would want to change the service parameter to pass the original party caller id through. I cannot think of another place this value would get leveraged. For AAR or SRST you always call the pilot. No devices every really try to call the vm ports themselves. Please let me know if there is some other feature that would make setting the VM port external number mask useful. Very curious as to the motivation to set this. thanks steve On Thu, Mar 28, 2013 at 7:43 AM, William Bell b...@ucguerrilla.com wrote: I follow the requirements for site A phones but I mask with the number of the VM pilot. So, using ipExpert lab examples: Site A would be +1202555 for the phones Unity Connection VM Pilot number is 2600 I set the external mask on vm ports to +12025552600. -BIll -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 27, 2013, at 9:50 PM, Barrera, Hugo wrote: Hi Guy’s, ** ** For the voicemail ports on SA what do you recommend to put for the external mask? Should it match the phones external mask OR should it be only 10 digits because you’re not supposed to send the 1 out of SA? Thoughts would be appreciated? ** ** *- Hugo* ** ** ___ 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 ___ 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] Lab Exam Speed Strategy
What has been the turning corner for me regarding speed is don't run around the tree. What I mean by this is when you come across something in the lab try to plan ahead and pre-configure as much as you can. For example when you go thru call routing in cucm don't just set up partitions and css's...pre-build your RP's, RL's, RG's, AAR grp's, calling party TP's, TP's... Really try to nail things and pre-config when you hit any section in the lab. What has also helped me is drilling notepad for IOS configurations as well. HTH Regards, Hugo From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of CCIEing Sent: Thursday, March 28, 2013 12:44 AM To: Jamie Parr (jamparr) Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Lab Exam Speed Strategy Hi Jamie, Would you please explain this more : You have to setup most devices with little or no prior configuration, there are things that cannot change. Know these things and practice them over and over so you do not have to think about them Thanks in advance On Wed, Mar 27, 2013 at 9:00 PM, Jamie Parr (jamparr) jamp...@cisco.commailto:jamp...@cisco.com wrote: First attempt I was very slow - did not use the device based approach, did not finish all tasks. Second I was much faster - using the device based approach, finished with 3 hours to test. Third attempt I finished with more than 3 hours to test and pick up the issues - Passed My advice: * I found the more I practiced the faster I got, practice practice practice * Use notepad to write all your device configs first, you can copy and paste large sections of config saving a lot of time * Do not be so strict to the device based approach, use it as a base and create your own hybrid * You have to setup most devices with little or no prior configuration, there are things that cannot change. Know these things and practice them over and over so you do not have to think about them * Persevere, it's not easy and it sucks most of the time but you will get there Hope this helps Jamie Parr CCIE #38633 (voice) Engineer - IT jamp...@cisco.commailto:jamp...@cisco.com Phone: +44 20 8824 2641 Mobile: +44 7590622049 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 Dane Warner Sent: 26 March 2013 23:41 To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Lab Exam Speed Strategy To All, I took my second attempt on Monday, March 25 and did not pass. I was hoping for some insight on concrete suggestions to get faster. I didn't get hung up on any one task, I seemed to keep moving forward and tried to type as fast as I could, using CLI shortcuts, etc. I used the device-based methodology and I feel pretty confident of my technical knowledge. Yet I didn't even get to many tasks at all, I would have needed another 2-3 hours to complete all tasks. I hear of candidates completing all tasks in 6-7 hours, which means I would need to become twice as fast as my last attempt. It almost sounds insurmountable. Do I need to take typing classes? Any recommendations that don't break the NDA would be greatly appreciated. Regards, Dane Warner, CCVP Sr. Network Engineer Epoch Universal, Inc. (909)226-0755 dwar...@epochuniversal.commailto:dwar...@epochuniversal.com [Epoch_Logo_Smaller_Transparent] ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.PlatinumPlacement.com inline: image001.png___ 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] Voicemail Ports on SA
Steve, Not sure if you were presenting that question to me or the OP. I can answer from my perspective: I set the emask on the ports because I feel it is best practice and it is part of my base config. I am already dorking with the VM ports to assign a CSS, AAR CSS, and AAR group and while I am there I mod the emask and save the config. As far as relevance or how it is used. If you didn't toggle the Display Original Calling Number on Transfer from Unity from the default value then the port emask would be used for direct and transfer calls from CUC. If you toggled the aforementioned service parameter to true then the emask on the port would only be used for direct calls. A relevant IE question involving direct calls would be sending notification messages to a telephone number. Whether that question has been / is / will be on an IE exam is any one's guess. Again, I set the emask to the vmpilot number as part of base config template. Now, if I wasn't already going into the VM port for another reason then I'd likely say screw it unless there was a question that made me deal with it. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 28, 2013, at 2:36 PM, Steve Keller wrote: Would this only be used if there is a call outbound from Unity and you do not have the service parameter to use original caller id when call routes through unity? Not sure of the exact parameter name , but i think everyone is familiar with that one by now. Thus caller id would be Voicemail/+12025552002 for a call the came from Unity. Even in this case i would think you would want to change the service parameter to pass the original party caller id through. I cannot think of another place this value would get leveraged. For AAR or SRST you always call the pilot. No devices every really try to call the vm ports themselves. Please let me know if there is some other feature that would make setting the VM port external number mask useful. Very curious as to the motivation to set this. thanks steve On Thu, Mar 28, 2013 at 7:43 AM, William Bell b...@ucguerrilla.com wrote: I follow the requirements for site A phones but I mask with the number of the VM pilot. So, using ipExpert lab examples: Site A would be +1202555 for the phones Unity Connection VM Pilot number is 2600 I set the external mask on vm ports to +12025552600. -BIll -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 27, 2013, at 9:50 PM, Barrera, Hugo wrote: Hi Guy’s, For the voicemail ports on SA what do you recommend to put for the external mask? Should it match the phones external mask OR should it be only 10 digits because you’re not supposed to send the 1 out of SA? Thoughts would be appreciated? - Hugo ___ 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 ___ 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] MOH Music On Hold source from local router issue
Dear Experts, I am trying to configure MOH in order to make SB Router source its music on hold from the local router. thats my configs A-Enable Multicast MOH for the audio stream: Go to CUCM ---Media Resources---Music On Hold Audo Source Tick play continuously , Allow Multicasting B-Enable Multicast MOH for the MOH server: Publisher will be unicast MOH Server for HQ Subscriber will be multicast for SB Site Go to Media Resources --- MOH Server---MOH_3(Subscriber) Make MOH Device Pool Enable Multicast Audo Source on this MOH Server C-Create a Media Resource Group (MRG) for unicast MOH: Media Resources Media Resource Group Add New Name: MOH_UNICAST Selected Media Resources: MOH_2 (MOH) Make sure use multicast in unticked D-Create a Media Resource Group (MRG) for multicast MOH: Media Resources Media Resource Group Add New Name: MOH_MCAST Selected Media Resources: MOH_3 (MOH) Make sure use multicast in ticked E-Assign the newly created MRGs to appropriate Media Resource Group List (MRGL): Under HQ MRGL Add MOH_UNICAST Under SB MRGL Add MOH_MCAST Then Reset Device Pool to take effect On your SRST make sure that you make multicast SB call-manager-fallback moh music-on-hold.au multicast moh 239.1.1.1 port 16384 route 142.1.65.254 142.102.65.254 loopback -to voice vlan exit ccm-manager music-on-hold ip multicast-routing int vlan 302 ip pim dense-mode int lo0 ip pim dense-mode I created a Region called MOH and it's G711 with all sites HQ , SB , SC When I call and place on hold and try to issue show ccm-manager music-on-hold i see 0 active calls knowing that also on the HQ Phone 1 and SB Phone 1 I put a music on hold source and without. All phones are in the correct Device pool , region and location. I have noticed it's beeping while on hold that means it's unable to invoke the moh knowing that i checked the flash: of the router it has the moh file correclty ___ 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] Voicemail Ports on SA
Thanks Bill. All of that makes perfect sense. I was just curious if there were something I was not considering about this. While yiu are there it probably doesn't hurt to set it. Just making sure I am not totally overlooking something. On Mar 28, 2013 3:27 PM, William Bell b...@ucguerrilla.com wrote: Steve, Not sure if you were presenting that question to me or the OP. I can answer from my perspective: I set the emask on the ports because I feel it is best practice and it is part of my base config. I am already dorking with the VM ports to assign a CSS, AAR CSS, and AAR group and while I am there I mod the emask and save the config. As far as relevance or how it is used. If you didn't toggle the Display Original Calling Number on Transfer from Unity from the default value then the port emask would be used for direct and transfer calls from CUC. If you toggled the aforementioned service parameter to true then the emask on the port would only be used for direct calls. A relevant IE question involving direct calls would be sending notification messages to a telephone number. Whether that question has been / is / will be on an IE exam is any one's guess. Again, I set the emask to the vmpilot number as part of base config template. Now, if I wasn't already going into the VM port for another reason then I'd likely say screw it unless there was a question that made me deal with it. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 28, 2013, at 2:36 PM, Steve Keller wrote: Would this only be used if there is a call outbound from Unity and you do not have the service parameter to use original caller id when call routes through unity? Not sure of the exact parameter name , but i think everyone is familiar with that one by now. Thus caller id would be Voicemail/+12025552002 for a call the came from Unity. Even in this case i would think you would want to change the service parameter to pass the original party caller id through. I cannot think of another place this value would get leveraged. For AAR or SRST you always call the pilot. No devices every really try to call the vm ports themselves. Please let me know if there is some other feature that would make setting the VM port external number mask useful. Very curious as to the motivation to set this. thanks steve On Thu, Mar 28, 2013 at 7:43 AM, William Bell b...@ucguerrilla.comwrote: I follow the requirements for site A phones but I mask with the number of the VM pilot. So, using ipExpert lab examples: Site A would be +1202555 for the phones Unity Connection VM Pilot number is 2600 I set the external mask on vm ports to +12025552600. -BIll -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 27, 2013, at 9:50 PM, Barrera, Hugo wrote: Hi Guy’s, ** ** For the voicemail ports on SA what do you recommend to put for the external mask? Should it match the phones external mask OR should it be only 10 digits because you’re not supposed to send the 1 out of SA? Thoughts would be appreciated? ** ** *- Hugo* ** ** ___ 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 ___ 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
