[OSL | CCIE_Voice] CCIE # 38487
All, I'm pleased to let everyone know that I got my numbers! I am CCIE # 38487. It's been a very long road. For me, my pursuit of the CCIE lasted a little over 2 years. I used IPExpert as my end to end solution for the CCIE. I went through the initial Volume 1 and 2 study material and then attended the 2 week bootcamp with Vik. I can truly say that the bootcamp was the piece of the puzzle that put everything together for me. I highly recommend it to everyone. While I didn't pass in 3 attempts, which was my goal, I'm happy to have finished. My advice to everyone is to make sure you know ALL the technology and can set it up efficiently and effectively. Furthermore, I think the key is perserverance. Everything probably won't go according to plan or your schedule. Don't let that phase you and when you get close book every 30 days until you clear it. For me I eventually switched to a hybrid device based approach. My strategy was to apply a base configuration to all devices in order that was repeatable based on what I was being asked. Once complete I would then start working through the lab in order, but at a much quicker pace as most of my configuration was in place. I want to thank everyone on this mailer for their continued support and knowledge sharing. I wish everyone luck! Thanks, Jason Lee ___ 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] Phone NTP Reference Vs NTP Reference
I had the same impression as Bill. It would be very interesting if that was the case. Sent from my iPhone On Mar 6, 2013, at 7:49 AM, William Bell b...@ucguerrilla.com wrote: Pixar, Are you certain about the Phone NTP reference and CUPC? I have not heard that before. I was under the impression that CUPC would use the clock of the underlying OS. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 6, 2013, at 12:15 AM, Pixar Perfect wrote: you still need the Phone NTP reference on the labs as CUPC client is a SIP client ..there are no SIP phones on the Version 3 labs but we might see lot on Version 4. Date: Tue, 5 Mar 2013 01:05:22 +0300 From: aboaz...@gmail.com To: corygray22...@hotmail.com; bring...@gmail.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Phone NTP Reference Vs NTP Reference Oh thanks a lot for your input. Appreciated .. On Tue, Mar 5, 2013 at 12:48 AM, Cory Gray corygray22...@hotmail.com wrote: Phone ntp reference is for SIP phones only Sent from my iPhone On Mar 4, 2013, at 4:42 PM, CCIEing aboaz...@gmail.com wrote: Hello All, The following question cross my mind while doing the NTP configuration stuff.. What is the difference between the Phone NTP reference configuration in the CCM Web administration page and The NTP reference on the OS Administration page?? does the 1st one for the endpoints where the 2nd one is for the CUCM itself? Thanks ___ 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 ___ 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] RSVP a big problem (sanity insanity)
Not sure it makes any difference in this situation, but I never use codec pass-through on my configuration. I've never had any issues. On Wed, Mar 6, 2013 at 12:32 PM, michael.se...@compucom.com wrote: --MJ Your problem is a misconfigured location somewhere in CUCM. Your configuration on the gateways is correct to allow 4 calls using RSVP based CAC. In my experience the issue your running into is not going to be an issue with the configuration on your gateways (use show SCCP on gateways to verify media resource registration), but a misconfigured location in CUCM of an assignment of a location either on phone, gateway or device pool. Not only are your calls not invoking CAC/AAR but they are NOT rerouting which points to your Route Patterns/Route List configuration. You might also verify the mask on your phones regarding AAR kicking in as well as applying the AAR calling search space on the gateways and the Device level of the phone. You also need to apply the AAR group to the gateway and Phone device level. On the live level you must also set the AAR group. Michael Sears CCIE (V) 38404 2. RSVP a big problem (sanity insanity) -- Message: 2 Date: Wed, 6 Mar 2013 21:49:54 +0530 From: sanity insanity networksanitytoinsan...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] RSVP a big problem Message-ID: cag4zmyw5dpqbxmgrcj3finope+pnur8zbjepv26cywgqyfh...@mail.gmail.com Content-Type: text/plain; charset=utf-8 hi Guys, I have to Configure IP Phones and gateways in such as way that all calls within same site should use G711 Codec. Also, all calls between the sites to remote IP phones and gateways should use G729 Codec. RSVP Call Admission Control (CAC) between HQ and branch site based on bandwidth limitations. There can be 4 concurrent calls. G711 CODEC to be used for multi-directional audio. Steps:- 1) I set the location Bw between my headquater and branch as Mandatory. 2) I also have the MTP registered and added to the correct MRG MRGLs 3) The following is a snip of my config on headquarter... dspfarm profile 1 mtp no codec g711u codec g729r8 codec pass?through rsvp maximum sessions software 4 associate application SCCP ! interface Serial0/0/0.2 point?to?point ip rsvp bandwidth 112 # 4 call similarly on branch site... dspfarm profile 1 mtp no codec g711u codec g729r8 codec pass?through rsvp maximum sessions software 4 associate application SCCP ! interface Serial0/0/0.2 point?to?point ip rsvp bandwidth 112 # 4 call Questions: == 1) With the above config I notice that when I make a call from headquarter site 2XXX to branch site 4XXX . The message on the phone is Not enough Bandwidth and the call disconnects. What is the exact problem? 2) Is my config above correct? -MJ ___ 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] Phone NTP Reference Vs NTP Reference
The Phone NTP Reference is used for SIP endpoints. SIP endpoints store a NTP server address internally and they use the Phone NTP Reference parameter to obtain that information. This parameter is not required for SCCP endpoints. The second is for the CUCM server. You were pretty much spot on with your guess. HTH, Jason On Mon, Mar 4, 2013 at 4:15 PM, CCIEing aboaz...@gmail.com wrote: Hello All, The following question cross my mind while doing the NTP configuration stuff.. What is the difference between the Phone NTP reference configuration in the CCM Web administration page and The NTP reference on the OS Administration page?? does the 1st one for the endpoints where the 2nd one is for the CUCM itself? Thanks ___ 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] SNR
The call to the mobile phone from the Mobility feature uses the Redirecting CSS at the RDP of the user. None of the internal phones would have access to use that CSS unless ofcourse it is the same CSS assigned to phones, but that wouldn't be a great way to set up SNR. It's best to have dedicated Class of Control. What is the behavior you get when you dial (Call Can't be Completed as Dialed or fast busy)? If you are getting the message about incorrectly dialed number I'd start looking at the Route Patterns, Partitions and Calling Search Spaces associated with the phones in CUCM. If it's fast busy you may have a case where you aren't passing the right digits or plan/type to the PSTN. Standard PSTN debugging should work to help out with that (debug voice diapleer and debug issn q931). Hope this helps! Jason On Sun, Mar 3, 2013 at 8:36 PM, Stacy Vacca stacy.va...@gmail.com wrote: I am working on SRN in Lab 5C. I am having issues with Internal Calls ringing my mobile device. I call from CME Site or PSTN. It successfully rings line 2 on PSTN Phone. I call from any phone registered to CUCM and it will not ring my mobile number on the PSTN Phone. I can answer the call on HQ Phone 2 and push the mobility button and successfully transfer the call out and then bring it back. Thoughts? Thanks Stacy ___ 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] Class-Based Shaping Issue
I reloaded the router and reapplied the QoS configuration. No dice. Still not matching in the nested policy. Very weird that it is working like a champ in the SB router. I've done this a couple times in the past and never run into this issue. I'm going to play with it a little longer, but may have to come back to it in the morning with a fresh set of coffee fueled eyes... On Sun, Mar 3, 2013 at 10:07 PM, Bill whl...@gmail.com wrote: Well it looks like it is picking up the RTP header compression, shaping and even the service policy but not doing it. Have you tried rebooting the router or copy, delete then reapply ? Sent from my iPad On Mar 3, 2013, at 8:43 PM, Jason Lee jas7...@gmail.com wrote: All, Was wondering if I could get a second set of eyes on my Class Based Shaping configuration. I just implemented between SA and SB. The same configuration is used on both. On SB I see the packets being captured in both policy maps. On SA I don't see traffic being captured in the nested policy-map containing voice and signaling configuration. Heres the output... r3800-2j-a(config)#do sh policy-map inter s0/2/0.1 Serial0/2/0.1: DLCI 201 - Service-policy output: shape-policy-map Class-map: class-default (match-any) 3234 packets, 563006 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) 0/0/0 (pkts output/bytes output) 3234/545846 shape (average) cir 384000, bc 3840, be 0 target shape rate 384000 lower bound cir 0, adapt to fecn 0 compress: header ip rtp UDP/RTP (compression on, Cisco, RTP) Sent:455 total, 452 compressed, 17160 bytes saved, 10140 bytes sent 2.69 efficiency improvement factor 99% hit ratio, five minute miss rate 0 misses/sec, 0 max rate 0 bps Service-policy : llq 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 r3800-2j-a(config)#do sh policy-map inter s0/2/0.1 Serial0/2/0.1: DLCI 201 - Service-policy output: shape-policy-map Class-map: class-default (match-any) 3234 packets, 563006 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) 0/0/0 (pkts output/bytes output) 3234/545846 shape (average) cir 384000, bc 3840, be 0 target shape rate 384000 lower bound cir 0, adapt to fecn 0 compress: header ip rtp UDP/RTP (compression on, Cisco, RTP) Sent:455 total, 452 compressed, 17160 bytes saved, 10140 bytes sent 2.69 efficiency improvement factor 99% hit ratio, five minute miss rate 0 misses/sec, 0 max rate 0 bps Service-policy : llq 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: voice (match-all) 0 packets, 0 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: ip dscp ef (46) Match: protocol rtp Match: protocol rtcp Priority: 24 kbps, burst bytes 1500, b/w exceed drops: 0 Class-map: signal (match-all) 0 packets, 0 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: ip dscp cs3 (24) Match: protocol sip Match: protocol h323 Match: protocol mgcp Match: protocol skinny Queueing queue limit 64 packets (queue depth/total drops/no-buffer drops) 0/0/0 (pkts output/bytes output) 0/0 bandwidth 18 kbps Class-map: class-default (match-any) 3234 packets, 563006 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: any queue limit 64 packets (queue depth/total drops/no-buffer drops) 0/0/0 (pkts output/bytes output) 3234/545846 r3800-2j-a(config)# 5 minute offered rate 0 bps, drop rate 0 bps Match: ip dscp ef (46) Match: protocol rtp Match: protocol rtcp Priority: 24 kbps, burst bytes 1500, b/w exceed drops: 0 *Class-map: signal (match-all)* * 0 packets, 0 bytes* 5 minute offered rate 0 bps, drop rate 0 bps Match: ip dscp cs3 (24) Match: protocol sip Match: protocol h323 Match: protocol mgcp Match: protocol skinny Queueing
Re: [OSL | CCIE_Voice] Unity Connection user template time format
Bill, I like your method. Haven't been setting time zone up to this point, but think I will going forward using your method. Thanks for the info! Sent from my iPhone On Feb 28, 2013, at 10:53 AM, William Bell b...@ucguerrilla.com wrote: First, I believe you do want the users provisioned in CUC to be provisioned with the correct timezone. Second, the method followed is up to you. I do the following: 1. Create hqusers template based on voicemailusers template. - Change timezone - Change tutorial option - Change password options (GUI and TUI) - Change password (GUI and TUI) 2. Create sbusers tmplate based on HQ - Change timezone 3. Create scusers template based on HQ - Change timezone Import users based on the appropriate template. The above is my preference. I see it this way. I have to dork with the templates anyway. I have to at least create one that modifies tutorial, password settings, etc. The other two templates only require one change each. So, that is changing two elements. In contrast, if I import all users using the same template then I have to possibly go to 4 users and make the same change. So, I am potentially changing four elements. Maybe one argues that it could be less than 4 elements (users). I don't care. At that point, it is more efficient for me to have a method that is more flexible and stick to it then ponder over such a small task at exam time. Just shoot and scoot. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Feb 28, 2013, at 9:09 AM, Cory Gray wrote: I would rather do it on the subscriber page vs changing the template multiple times. I think that would be faster but as always, go with whatever you practice. *From:* Chrysostomos Christofi [mailto:ch.christ...@logicom.netch.christ...@logicom.net ] *Sent:* Thursday, February 28, 2013 9:07 AM *To:* Cory Gray; 'Nicolas MICHEL'; 'Jamie Parr (jamparr)' *Cc:* ccie_voice@onlinestudylist.com *Subject:* RE: [OSL | CCIE_Voice] Unity Connection user template time format Guys Take it logically If HQ site has different time zone with Site B then for sure the users in CUC must have the correct time zone for each branch 1) User template in CUC (modify there anything you want include time zone),Import HQ users 2) Then modify again the user template to the correct time zone for users in site B and then import the users for site B Regards *From:* ccie_voice-boun...@onlinestudylist.com [ mailto:ccie_voice-boun...@onlinestudylist.comccie_voice-boun...@onlinestudylist.com ] *On Behalf Of *Cory Gray *Sent:* Πέμπτη, 28 Φεβρουαρίου 2013 2:53 μμ *To:* 'Nicolas MICHEL'; 'Jamie Parr (jamparr)' *Cc:* ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] Unity Connection user template time format I had struggled with whether to match each subscriber with their correct time zone. My GUESS is that it only matters if a Unity Connection question involves any type of time stamp such as when the message was delivered. It probably cannot hurt to do it as a best practice as I seriously doubt it can hurt your scoring but you never know so you have to decide what is best. *From:* ccie_voice-boun...@onlinestudylist.com [ mailto:ccie_voice-boun...@onlinestudylist.comccie_voice-boun...@onlinestudylist.com ] *On Behalf Of *Nicolas MICHEL *Sent:* Thursday, February 28, 2013 6:45 AM *To:* Jamie Parr (jamparr) *Cc:* ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] Unity Connection user template time format Does CUCN has something to do with the display of the phone ? :=) Le 2/28/2013 12:07 PM, Jamie Parr (jamparr) a écrit : Hi all If we are instructed to display the phones time in 24 hour format, should we reflect this in the user templates for Cisco Unity? Thanks image001.jpg *Jamie Parr* Engineer - IT jamp...@cisco.com Phone: *+44 20 8824 2641* Mobile: *+44 7590622049* *Cisco Systems* 9-11 New Square Bedfont Lakes Feltham Middlesex TW14 8HA United Kingdom www.cisco.com http://cisco.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 ___ 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] Questions regarding Unified CME as SRST
Quick update on my strategy... I've usually shied away from pre-configuring SRST as it tends to register my phones prior to them being ready to register to CUCM while I'm doing my base lab setup. Probably not a big deal, but it's just one of those things that irks me. To get around that I've started pre-configuring SRST again, but not including the ip source command. This allows me to get my entire config in the router, but not initiate a failover until I'm ready. When I get to the HA portion of a lab, I then add the source command in. Does this seem like a good strategy? On Thu, Feb 21, 2013 at 3:56 PM, Jason Lee jas7...@gmail.com wrote: Sounds good to me. I'm probably in a little late in the game to change my strategy now, but if you can make it work for you it sounds like it could be very beneficial. I can't say I've run into any issues with the preconfigured templates though. What have you seen? On Thu, Feb 21, 2013 at 4:51 AM, Bill whl...@gmail.com wrote: Hi Bill Read the question carefully but if you can control the config it is better than trusting something you don't trust Bill Sent from my iPad On Feb 21, 2013, at 12:29 AM, William Bell b...@ucguerrilla.com wrote: Leslie/Steve/Jason, What are your thoughts on pre-configuring ephone-dns when you are permitted to use CME-SRST with autoprovision dn or all? Instead of dorking around with templates (which I hear is flaky) I was thinking about tweaking my approach to pre-configure ephone-dns when I build out SRST. I have done some basic tests and read the docs. It is supported and appears to work. The benefits: I don't have to wait for phones to failover to finish SRST related configs. I can configure BACD, call coverage for VM, mwi sip, name, description, etc. Thoughts? -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Feb 20, 2013, at 10:23 PM, Leslie Meade wrote: Hey Steve, ** ** I just ran this via my lab and the light turns on.. If I run debug ccsip messages I see the cue send a mwi notify to the ephone and the light comes on ** ** ** ** R3(config)# *Feb 21 03:11:56.231: %IPPHONE-6-REG_ALARM: 10: Name=SEP001BD4607B13 Load= SCCP41.8-4-1S Last=TCP-timeout *Feb 21 03:11:56.279: %IPPHONE-6-REGISTER: ephone-2:SEP001BD4607B13 IP:10.69.66.20 Socket:1 DeviceType:Phone has registered. *Feb 21 03:11:58.615: %IPPHONE-6-REG_ALARM: 10: Name=SEP0017E066C2E7 Load= SCCP41.8-4-1S Last=TCP-timeout *Feb 21 03:11:58.679: %IPPHONE-6-REGISTER: ephone-1:SEP0017E066C2E7 IP:10.69.66.21 Socket:2 DeviceType:Phone has registered. *Feb 21 03:12:15.235: //-1//SIP/Msg/ccsipDisplayMsg: Received: NOTIFY sip:4002@10.69.66.254:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.69.66.253:5060 ;branch=z9hG4bKyHFdoT6xNYZ85fvOD9z4kQ~~1 Max-Forwards: 70 To: sip:4002@10.69.66.254:5060 From: sip:4002@10.69.66.253:5060;tag=ds3be1f82d Call-ID: c234c79-1100@sip:4002@10.69.66.253:5060 CSeq: 1 NOTIFY Content-Length: 115 Contact: sip:4002@10.69.66.253:5060 Content-Type: application/simple-message-summary Event: message-summary ** ** Messages-Waiting: yes Message-Account: sip:4002@10.69.66.253 Voice-Message: 1/0 (0/0) Fax-Message: 0/0 (0/0) ** ** *Feb 21 03:12:15.243: //-1//SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.69.66.253:5060 ;branch=z9hG4bKyHFdoT6xNYZ85fvOD9z4kQ~~1 From: sip:4002@10.69.66.253:5060;tag=ds3be1f82d To: sip:4002@10.69.66.254:5060;tag=3BF3A8-1459 Date: Thu, 21 Feb 2013 03:12:15 GMt Call-ID: c234c79-1100@sip:4002@10.69.66.253:5060 CSeq: 1 NOTIFY Content-Length: 0 ** ** ** ** ** ** sip-ua mwi-server ipv4:10.69.66.253 expires 3600 port 5060 transport udp unsolicited ! ! ! gatekeeper shutdown ! ! telephony-service srst mode auto-provision all srst ephone template 1 srst dn template 1 srst dn line-mode octo max-ephones 30 max-dn 30 no-reg both ip source-address 10.69.66.254 port 2000 time-zone 42 voicemail 4220 mwi relay max-conferences 8 gain -6 transfer-system full-consult transfer-pattern .T secondary-dialtone 9 create cnf-files version-stamp Jan 01 2002 00:00:00 ** ** *From:* ccie_voice-boun...@onlinestudylist.com [ mailto:ccie_voice-boun...@onlinestudylist.comccie_voice-boun...@onlinestudylist.com ] *On Behalf Of *Steve Keller *Sent:* Wednesday, February 20, 2013 12:23 PM *To:* Jason Lee *Cc:* ccie_voice *Subject:* Re: [OSL | CCIE_Voice] Questions regarding Unified CME as SRST ** ** Well i confirmed today that if using a CUCM-CUE integration at a branch site, th you will want to setup your MWI to be subscribe/notify when you complete your CUE integratoin with CUCM. MWI works great when
Re: [OSL | CCIE_Voice] B Channel Busy Out
I use this as a strategy for checking my gateway configuration Ensure that your are meeting requirements on the following 1. display-ie 2. BCHAN order selection (Ascending, Descending) 3. BCHAN number 1. How many BCHANs? 1. If not specified create a full PRI. 2. If fractional 1. Set BCHAN Maintenance in Advanced Service Parameters 2. Check the *Check Status *checkbox in GW config 4. Clocking 1. Network clock participate 2. network clock select 1 t10/0/0 5. ISDN Switch-Type 6. Source-Address 7. 911 1. Done in gateway section. 1. Make sure to have routed correctly, SLRG?, 8. Direct Inward Dial 9. clear counters On Wed, Feb 27, 2013 at 3:38 AM, Jamie Parr (jamparr) jamp...@cisco.comwrote: I am also curious as to the grading on the gateways, I received very low marks on this section. Can anyone help? ** ** Thanks ** ** *Jamie Parr* 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 *Pixar Perfect *Sent:* 26 February 2013 19:35 *To:* Steve Keller; GARY CLARK *Cc:* CCIE Voice OSL *Subject:* [OSL | CCIE_Voice] B Channel Busy Out ** ** Gary ..you mentioned B channel busyout on service parameter. in my understanding this was only needed when you would download the GW config from CCM i.e., ccm-manager config. it doesn't make any sense to use this service parameter as most of the solution guides (INE, IPXEPERT, 360) do not encourage the use of ccm-manager config except initial stage of your config and then disable it. I have heard ppl who passed just using standard configs but not sure if they did the B channel busy out on service parameter. ** ** ** ** mgcp mgcp call-agent 10.10.210.11 --sub mgcp dtmf mgcp bind ... (2x2) ** ** ccm-mana fall ccm-mana music ccm-mana mgcp ccm-mana red 10.10.210.10 -- pub ** ** ** ** if B channel status is *really graded *on the exam then it is one of those things that doesn't make sense to have it there but is needed to score points [image: Emoji] ** ** experts, any comments or advise from the recent Experts ? ** ** ** ** PIXAR ** ** ** ** -- Date: Mon, 25 Feb 2013 14:31:12 -0500 From: skeller...@gmail.com To: garyclark...@gmail.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] lab7 failed for 1% I recieved 29% in RTP on GW Signalling section and Call Routing as well. I am very discouraged i could score very low marks on these sections as i took my time and felt like i had nailed them. I scored really well in all other areas but failed because of these 2 sections. It is a mystery to me what the proctor is doing to arrive at that score, when all my calls worked, the debugs matched the requirements, i was binding to the correct interfaces, setting up the correct protocol and channels,etc. I would love to hear what insight folks have as to why the scores could be so low when everything looked to be working beautifully, without breaking NDA of course. thanks steve On Mon, Feb 25, 2013 at 1:54 PM, GARY CLARK garyclark...@gmail.com wrote: Hi friends, I got lab 7 in SJC and i completed that in 6 hrs and tested for 2 hrs time. I thought i have passed 1000% but when i saw my result i was surprised.*** * I almost got everywhere 100% except VG / 29% which was 17 marks section.** ** Same story with my friends do anyone got 100% in VG for lab 7 If anyone interested to share the hidden secrets then welcome as people are getting lab 7 repeating now very eager to understand what could be wrong. Please email me for further discussion. We 3 friends attempted out of which i also did busy out channel but that also did not helped its 29% only why so ** ** Regards ___ 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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE
Re: [OSL | CCIE_Voice] Questions regarding Unified CME as SRST
Sounds good to me. I'm probably in a little late in the game to change my strategy now, but if you can make it work for you it sounds like it could be very beneficial. I can't say I've run into any issues with the preconfigured templates though. What have you seen? On Thu, Feb 21, 2013 at 4:51 AM, Bill whl...@gmail.com wrote: Hi Bill Read the question carefully but if you can control the config it is better than trusting something you don't trust Bill Sent from my iPad On Feb 21, 2013, at 12:29 AM, William Bell b...@ucguerrilla.com wrote: Leslie/Steve/Jason, What are your thoughts on pre-configuring ephone-dns when you are permitted to use CME-SRST with autoprovision dn or all? Instead of dorking around with templates (which I hear is flaky) I was thinking about tweaking my approach to pre-configure ephone-dns when I build out SRST. I have done some basic tests and read the docs. It is supported and appears to work. The benefits: I don't have to wait for phones to failover to finish SRST related configs. I can configure BACD, call coverage for VM, mwi sip, name, description, etc. Thoughts? -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Feb 20, 2013, at 10:23 PM, Leslie Meade wrote: Hey Steve, ** ** I just ran this via my lab and the light turns on.. If I run debug ccsip messages I see the cue send a mwi notify to the ephone and the light comes on ** ** ** ** R3(config)# *Feb 21 03:11:56.231: %IPPHONE-6-REG_ALARM: 10: Name=SEP001BD4607B13 Load= SCCP41.8-4-1S Last=TCP-timeout *Feb 21 03:11:56.279: %IPPHONE-6-REGISTER: ephone-2:SEP001BD4607B13 IP:10.69.66.20 Socket:1 DeviceType:Phone has registered. *Feb 21 03:11:58.615: %IPPHONE-6-REG_ALARM: 10: Name=SEP0017E066C2E7 Load= SCCP41.8-4-1S Last=TCP-timeout *Feb 21 03:11:58.679: %IPPHONE-6-REGISTER: ephone-1:SEP0017E066C2E7 IP:10.69.66.21 Socket:2 DeviceType:Phone has registered. *Feb 21 03:12:15.235: //-1//SIP/Msg/ccsipDisplayMsg: Received: NOTIFY sip:4002@10.69.66.254:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.69.66.253:5060;branch=z9hG4bKyHFdoT6xNYZ85fvOD9z4kQ~~1 Max-Forwards: 70 To: sip:4002@10.69.66.254:5060 From: sip:4002@10.69.66.253:5060;tag=ds3be1f82d Call-ID: c234c79-1100@sip:4002@10.69.66.253:5060 CSeq: 1 NOTIFY Content-Length: 115 Contact: sip:4002@10.69.66.253:5060 Content-Type: application/simple-message-summary Event: message-summary ** ** Messages-Waiting: yes Message-Account: sip:4002@10.69.66.253 Voice-Message: 1/0 (0/0) Fax-Message: 0/0 (0/0) ** ** *Feb 21 03:12:15.243: //-1//SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.69.66.253:5060;branch=z9hG4bKyHFdoT6xNYZ85fvOD9z4kQ~~1 From: sip:4002@10.69.66.253:5060;tag=ds3be1f82d To: sip:4002@10.69.66.254:5060;tag=3BF3A8-1459 Date: Thu, 21 Feb 2013 03:12:15 GMt Call-ID: c234c79-1100@sip:4002@10.69.66.253:5060 CSeq: 1 NOTIFY Content-Length: 0 ** ** ** ** ** ** sip-ua mwi-server ipv4:10.69.66.253 expires 3600 port 5060 transport udp unsolicited ! ! ! gatekeeper shutdown ! ! telephony-service srst mode auto-provision all srst ephone template 1 srst dn template 1 srst dn line-mode octo max-ephones 30 max-dn 30 no-reg both ip source-address 10.69.66.254 port 2000 time-zone 42 voicemail 4220 mwi relay max-conferences 8 gain -6 transfer-system full-consult transfer-pattern .T secondary-dialtone 9 create cnf-files version-stamp Jan 01 2002 00:00:00 ** ** *From:* ccie_voice-boun...@onlinestudylist.com [ mailto:ccie_voice-boun...@onlinestudylist.comccie_voice-boun...@onlinestudylist.com ] *On Behalf Of *Steve Keller *Sent:* Wednesday, February 20, 2013 12:23 PM *To:* Jason Lee *Cc:* ccie_voice *Subject:* Re: [OSL | CCIE_Voice] Questions regarding Unified CME as SRST* *** ** ** Well i confirmed today that if using a CUCM-CUE integration at a branch site, th you will want to setup your MWI to be subscribe/notify when you complete your CUE integratoin with CUCM. MWI works great when registered to CUCM and using CUE for VM. When the site fails over in to srst mode and your phone has an existing MWI on it, this is what you would want to do in order to preserve that MWI lamp. 1) When integrating your CUE to CUCM choose MWI type subscribe/notify. 2) When building your router config for SRST, make sure to build an ephone-dn-template that specifies MWI SIP that will get applied to the phones when they register (under your telephony service). 3) when configuring sip-ua / mwi-server i did not use unsolicited key word this has allowed the current MWI lamp to stay lit when failover to unified
Re: [OSL | CCIE_Voice] MWI Best Practice
I'm using unsolicited as well, unless I'm specifically asked or nudged by the exam to use a different method. Best to be prepared for all of options. I've never had any issues with unsolicited stability. On Wed, Feb 20, 2013 at 8:51 AM, Cory Gray corygray22...@hotmail.comwrote: I use unsolicited for both. Of course I do not know whether it is right or not though. ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Pixar Perfect *Sent:* Tuesday, February 19, 2013 11:06 PM *To:* CCIE Voice OSL *Subject:* [OSL | CCIE_Voice] MWI Best Practice ** ** Experts and wannabe experts friends, ** ** what are the best practices for MWI in CME and SRST modes for the CUE site BR2? i was used to using MWI ON and MWI OFF DNs on a CME but i was told by a fellow aspirant that MWI ON/OFF are not preferred (grading wise) and that solicited MWI is that gets you to the needed points. ** ** however i have seen solicited and unsolicited to be verify unreliable on 7965 phones .. you have to do no mwi sip and mwi sip to get solicited to work and sometimes reboot CUE or router to get both solicited and unsolicited to work. I am 1 month away from exam date and dont want to waste time exploring instead adopt best common practice that works flawlessly ..and so far it has been ON/OFF DN ___ 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] SRST issues
1. I can confirm that I also don't see type added. I don't seem to be suffering the side effect of the VM button not working though. I assume it's not required for that functionality in SRST? 2. I don't have Fallback configured currently, but I believe it auto populates the E164 as the name. Not sure there's a way to customize that in traditional SRST? To preserve calls try: To preserve h323 calls you need to do the following to things: * * *From the gateway:* voice service voip h323 call preserve ! Call preserve allows the call to be sustained if all connectivity to UCM servers is down. SRST. * * *In CUCM:* Go to CUCM Advanced Service parameters. Set the *Allow Peer to Preserve H323 Calls* * * 3. I think MVA is only supported by scripts native to CUCM. If your connectivity to CUCM is down, you can't access the MVA service in CUCM. I'm pretty sure there isn't native support of MVA in the router itself. It just points to the service in CUCM. 4. These are options I use to manipulate ANI into CUC. There was a good discussion on this about a month back on this mailer. *Alternate Extension in CUC* * * On the Unity Connections server, under the users mailbox configuration, you can set an alternate extension This should be set to the ANI (Calling number) of the phone calling into voicemail when calling through the PSTN. You can check debug isdn q931 on the SA router to determine what the inbound ANI is.* * * * *Calling Party Transformation Pattern on SA Device Pool* Preferred method when Alternate Extension is not allowed, because unlike the masking at the Hunt Pilot this preserves ANI for all other callers. Create a dedicated Partition and CSS for VM ANI manipulation. Then create a Calling Party Transformation pattern that strips the ANI of inbound calls to 4 digits and place it in the previously created Partition. I try to be as specific as possible here to meet the requirement for voicemail preservation. That way I'm not overlapping with other things. You can check debug isdn q931 on the SA router to determine what the inbound ANI is. Apply the CSS to the the SA Device Pool at the Calling Party Transformation CSS field *Calling Party Transformation Mask on VM Hunt Pilot* * * Easiest to configure, but manipulates all inbound ANI to CUC. If you have a requirement to playback the calling party ANI it will only list 4 digits. Under the Hunt Pilot that you setup to reach your voicemail, you can set the Calling Party Transformation Mask to (This will then only send 4 digits to the Unity system for Calling number no matter where the call comes in from) On Mon, Feb 18, 2013 at 11:25 PM, Ramcharan Arya ramcharan.a...@gmail.comwrote: Hello, I have couple of issues when site becomes SRST. 1. One ephone 1 and 2 it does not add phone type automatically. I'm using one 7962 and one 7965 phone. ccm-manager mgcp-fallback telephony-service srst mode auto all 2. My other site is H.323 gateway and I am using call-manager-fallback First of all my ie display showing E.164 number when making call to PSTN 2nd issue I am unable to preserve active call on H.323 gateway also keys are not same when phone register to cucm This site has mobile voice access it does not work. Is there any solution to change vxml script to use fall-manager-fallback feature. 4th issue : When I press message button I am able to reach mailbox using alternate ext in Unity. Is there any other approach if I am not allow to use alternate ext. Please help me how should I fix these problems. Thanks Regards, Ramcharan Arya CCIE # 28926 ___ 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] Questions regarding Unified CME as SRST
Steve, I think that if you set up Subscribe Notify for MWI instead of Unsolicited Notify it might preserve the light. In order to get that to work you would have had to load the phones into SRST (auto provision all) at least once so that they populate the running config. You can then configure the mwi sip option under the ephone-dn. That will force it to subsribe to to the CUE for MWI updates. I imagine that subscription happens every time the phone comes online or in this case when they register to the CME-SRST router during failover. It should then be followed by a notify with the MWI status. I did this on a straight CME lab yesterday and pulled the following traces. Given that occurs every time the phone boots up, you should meet your requirement. I'll test tomorrow since I'll be doing a 3 CUCM site lab. r2800-2j-b(config-ephone-dn)#mwi sip r2800-2j-b(config-ephone-dn)# Feb 18 21:11:30.316: //-1//SIP/Msg/ccsipDisplayMsg: Sent: SUBSCRIBE sip:3002@192.168.106.2:5060 SIP/2.0 -- HERE IS THE SUBSCRIBE MESSAGE Via: SIP/2.0/UDP 192.168.106.1:5060;branch=z9hG4bK191EA4 From: sip:3002@192.168.106.1;tag=58ACCE8-1615 To: sip:3002@192.168.106.2 Call-ID: 996B6A71-794611E2-80CEE1A3-4F484EB8@192.168.106.1 CSeq: 101 SUBSCRIBE Max-Forwards: 70 Date: Mon, 18 Feb 2013 21:11:30 GMT User-Agent: Cisco-SIPGateway/IOS-12.x Event: message-summary Expires: 3600 Contact: sip:3002@192.168.106.1:5060 Accept: application/simple-message-summary Content-Length: 0 Feb 18 21:13:11.067: //-1//SIP/Msg/ccsipDisplayMsg: Received: NOTIFY sip:3002@192.168.106.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.106.2:5060;branch=z9hG4bKUNWTUT.iNZVt5tr6uAHS+A~~3 Max-Forwards: 70 To: sip:3002@192.168.106.1;tag=58ACCE8-1615 From: sip:3002@192.168.106.2;tag=dec1fdb9-1100 Call-ID: 996B6A71-794611E2-80CEE1A3-4F484EB8@192.168.106.1 CSeq: 3 NOTIFY Content-Length: 114 Contact: sip:3002@192.168.106.2 Event: message-summary Allow-Events: refer Allow-Events: telephone-event Allow-Events: message-summary Subscription-State: active Content-Type: application/simple-message-summary Messages-Waiting: yes - HERE'S THE NOTIFICATION OF MWI ON Message-Account: sip:3002@192.168.106.2 Voice-Message: 1/0 (0/0) Fax-Message: 0/0 (0/0) On Tue, Feb 19, 2013 at 4:33 PM, Steve Keller skeller...@gmail.com wrote: Recently i have noticed a few things in my lab as i have been preparing for the lab exam. Using CME as SRST specifically in this situation, i have been trying to preserve as much features and appearance as i can when my UCM phones register to the gateway. Two scenarios i have question on because i cannot seem to get them to work. 1) If my branch 2 phone has a voicemail and MWI turned on, when it falls back to Unified CME as SRST the MWI does goes off, however i can retreive the vm because my CUE integratoin does remain in tact. Is it possible to have the phones fail over and maintain the MWI status automatically? If i leave a new vm while in SRST mode then the light does come on. 2)When making a call from the UCM phones to my BR2 phones (CFUR) the call comes in and at the gateway level i see the ANI is full e164 format including the + character. However the phone never shows the plus character in SRST mode. Is this possible? Does Unified CME as SRST support the + character? I am thinking if this is possible it would be nice to include these capabilities as part of my config if asked to preserve features, functionality while in SRST. thanks in advance all. ___ 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] Questions regarding Unified CME as SRST
I typically use unsolicited on my SRST sites for MWI, but you may be on to something. Maybe this method would be preferred. All depends what they are looking for! Thus begins my tangent ;o) I've seen the same behavior with the + as Bill. Sent from my iPhone On Feb 19, 2013, at 9:55 PM, William Bell b...@ucguerrilla.com wrote: Steve, Jason's response is spot on for your first question. Though, I have found the integration to be a little flaky myself. But that was a recent observation when I was trying pre-build ephone-dns before swinging a site to CME. In regards to your second question, I don't think the phone is display the + on the call plane. But it should display it in the status line at the bottom of the screen. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Feb 19, 2013, at 4:33 PM, Steve Keller wrote: Recently i have noticed a few things in my lab as i have been preparing for the lab exam. Using CME as SRST specifically in this situation, i have been trying to preserve as much features and appearance as i can when my UCM phones register to the gateway. Two scenarios i have question on because i cannot seem to get them to work. 1) If my branch 2 phone has a voicemail and MWI turned on, when it falls back to Unified CME as SRST the MWI does goes off, however i can retreive the vm because my CUE integratoin does remain in tact. Is it possible to have the phones fail over and maintain the MWI status automatically? If i leave a new vm while in SRST mode then the light does come on. 2)When making a call from the UCM phones to my BR2 phones (CFUR) the call comes in and at the gateway level i see the ANI is full e164 format including the + character. However the phone never shows the plus character in SRST mode. Is this possible? Does Unified CME as SRST support the + character? I am thinking if this is possible it would be nice to include these capabilities as part of my config if asked to preserve features, functionality while in SRST. thanks in advance all. ___ 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] Custom Tones
I ran through all the same scenarios as you and got the exact same results. I agree with your assessment of the situation. Lets hope we don't have to have a discussion with the Proctor about bogus requirements! Thanks so much for taking the time to give this the run through. On Mon, Feb 18, 2013 at 1:25 PM, William Bell b...@ucguerrilla.com wrote: Jason, I played with this some today and I think a lightbulb went off for me. The assumed scenario for cBarge + custom-cptone is: 1. PhoneC calls shared line on PhoneA/PhoneB (Phones A and B are registered to CME) 2. Phone A answers on shared line 3. Phone B seizes line (remote in use) and selects the cBarge softkey 4. At this point the custom-cptone for JOIN should be played out 5. Phone B disconnects from call 6. Our assumption is that the custom-cptone for LEAVE should be played out I have always had the same experience you noted. Which is: Step 4 works fine, no problem. Step 6 never works. IOW, I never hear a leave tone. I tested different configs for custom-cptone, even though doing so didn't make much sense. The behavior is the same. You do want to make sure that the frequency is different. The cadence can be the same as far as I can tell, but it can be diff too. Not really all that relevant to the question. I then tested MML using the same cptone setup and I do get JOIN and LEAVE tones. A clue that the voice-class assignment to the dspfarm is healthy. I then tested ad-hoc conference from one of the phones. Only test 3 party conference. I hear a JOIN tone when the 3rd party is added. I DO NOT hear a LEAVE tone when that third party disconnects. At this point it dawns on me what is going on. For giggles, I did another set of tests. I tested ad-hoc with 4 parties. I also tested a barge-in and then an ad-hoc add for a fourth party. If any single party (save the initiator) leaves that ad-hoc conference, a tone is played out to remaining parties (which is now 3). If one of the remaining three parties leaves (except for the conference initiator) then there is NO tone played out to the remaining two parties. Based on observed behavior, I am thinking that things are behaving as designed. The custom-cptones are associated with the dspfarm profile. When you transition from a 2-party call to a 3+ party call, you are involving the dspfarm and getting the tones. When you drop to a 2-party call, you are dropping the need for a dspfarm and the call becomes point-to-point. So, if the dspfarm was attempting to playout tones, it is no longer involved in the media path. So, the absence of the LEAVE tone seams (IMO) to be expected behavior. Assuming that one accepts that the observed behavior is expected then the question requirement to playout a tone when a party leaves is bogus. If I hit this in the real lab and the requirement says a tone must be played when the line is barged AND when the barging party leaves, I would bring it up to the proctor as a bogus requirement. The dspfarm is removed from the call at the point where the barging party leaves and is no longer in the media path. If, on the other hand, it simply says parties on the call should hear a tone when the line is barged then there is no problem. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Feb 17, 2013, at 9:45 PM, Jason Lee wrote: I'll give it a go tomorrow. I already reverted my pod this evening. I'll be doing another lab tomorrow, so I should be able to test this put by tomorrow afternoon. Sent from my iPad On Feb 17, 2013, at 9:14 PM, Bill whl...@gmail.com wrote: I think Justin might be on to it but it has been a while since I have done this in the lab. Sent from my iPad On Feb 17, 2013, at 3:06 PM, Justin Carney justin.s.car...@gmail.com wrote: I haven't tested this recently, but it may help to make the join/leave tones use different frequencies, as well as using different time intervals for the cadence. I'm not sure why you're getting these strange results (two tones on join when your cadence only shows one and no tone on leave), but there may be some strange feature (or bug) that has to do with both join and leave using the same frequency. voice class custom-cptone leave dualtone conference frequency 300 cadence 400 500 600 ! voice class custom-cptone join dualtone conference frequency 700 cadence 800 -Justin On Sun, Feb 17, 2013 at 1:56 PM, William Bell b...@ucguerrilla.comwrote: I don't have an answer for you. However, I can confirm that I have noticed the same behavior. When I have associated custom tones for join/leave events, I only hear the tone on join. Nada on leave. I haven't figured it out yet. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Feb 17, 2013, at 12:39 PM, Jason Lee wrote: All, I have continually struggled with custom tones for a while now. I'm working on the 5LB Lab 1
Re: [OSL | CCIE_Voice] CUE Dropped Calls
-net5 isdn incoming-voice voice isdn bchan-number-order ascending isdn outgoing display-ie no cdp enable ! interface Vlan1 no ip address ! router ospf 1 log-adjacency-changes network 192.168.0.0 0.0.255.255 area 0 ! ip forward-protocol nd ip route 0.0.0.0 0.0.0.0 Serial0/0/0.1 ip route 192.168.106.2 255.255.255.255 Service-Engine0/0 ip http server no ip http secure-server ! ! ! nls resp-timeout 1 cpd cr-id 1 ! ! ! ! ! ! control-plane ! ! ! voice-port 0/1/0:15 translation-profile incoming strip ! voice-port 0/2/0 ! voice-port 0/2/1 ! voice-port 0/2/2 ! voice-port 0/2/3 ! ! ! sccp local Loopback0 sccp ccm 192.168.106.1 identifier 3 version 7.0 sccp ccm 192.168.100.100 identifier 2 version 7.0 sccp ccm 192.168.100.101 identifier 1 version 7.0 sccp ! sccp ccm group 1 bind interface Loopback0 associate ccm 1 priority 1 associate ccm 2 priority 2 associate ccm 3 priority 3 associate profile 3 register sc-mtp-rsvp associate profile 2 register sc-conf associate profile 1 register sc-xcode keepalive timeout 3 switchover method immediate switchback method immediate ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 4 associate application SCCP ! dspfarm profile 2 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 2 conference-join custom-cptone join conference-leave custom-cptone leave associate application SCCP ! dspfarm profile 3 mtp codec g729r8 rsvp maximum sessions software 100 associate application SCCP ! ! dial-peer voice 911 pots translation-profile outgoing 911 destination-pattern 999$ port 0/1/0:15 forward-digits all ! dial-peer voice 97 pots translation-profile outgoing 97 destination-pattern 9[1-9]...$ port 0/1/0:15 forward-digits 8 ! dial-peer voice 910 pots translation-profile outgoing 910 destination-pattern 91[2-9].$ port 0/1/0:15 forward-digits 11 ! dial-peer voice 9011 pots translation-profile outgoing 9011 destination-pattern 900T port 0/1/0:15 prefix 00 ! dial-peer voice 1 pots incoming called-number . direct-inward-dial ! dial-peer voice 2 voip translation-profile incoming 9 destination-pattern 4...$ voice-class codec 1 voice-class h323 1 session target ipv4:192.168.100.101 incoming called-number . dtmf-relay h245-alphanumeric no vad ! dial-peer voice 3 voip preference 1 destination-pattern 4...$ voice-class codec 1 voice-class h323 1 session target ipv4:192.168.100.100 dtmf-relay h245-alphanumeric no vad ! dial-peer voice 999 pots ! dial-peer voice 4600 voip destination-pattern 4600$ session protocol sipv2 session target ipv4:192.168.106.2 dtmf-relay sip-notify codec g711ulaw no vad ! dial-peer voice 23 pots translation-profile outgoing 23 destination-pattern [23]...$ port 0/1/0:15 forward-digits all ! ! sip-ua mwi-server ipv4:192.168.106.2 expires 3600 port 5060 transport udp unsolicited ! ! ! gatekeeper shutdown ! ! telephony-service sdspfarm units 3 conference hardware srst mode auto-provision all srst ephone template 1 srst dn template 1 srst dn line-mode octo max-ephones 25 max-dn 40 no-reg ip source-address 192.168.106.1 port 2000 time-zone 21 time-format 24 voicemail 4600 max-conferences 8 gain -6 transfer-system full-consult secondary-dialtone 9 create cnf-files version-stamp 7960 Feb 16 2013 20:47:40 ! ! ephone-dn-template 1 call-forward busy 4600 call-forward noan 4600 timeout 10 ! ! ephone-template 1 softkeys remote-in-use Newcall CBarge ! ! ephone-dn 1 octo-line number 4001 description +442077964001 name +442077964001 ephone-dn-template 1 ! ! ephone-dn 2 octo-line number 4000 description 4000 name 4000 ephone-dn-template 1 ! ! ephone-dn 3 octo-line number 4002 description +442077964002 name +442077964002 ephone-dn-template 1 ! ! ephone-dn 20 octo-line number A02 no-reg primary conference ad-hoc ! ! ephone 1 privacy off privacy-button device-security-mode none mac-address C8F9.F9D7.545D ephone-template 1 button 1:1 2:2 ! ! ! ephone 2 device-security-mode none mac-address C8F9.F9D7.3977 ephone-template 1 button 1:3 2:2 ! ! ! line con 0 line aux 0 line 194 no activation-character no exec transport preferred none transport input all transport output pad telnet rlogin lapb-ta mop udptn v120 ssh line vty 0 4 password cisco login ! scheduler allocate 2 1000 ntp server 192.168.96.10 end r2800-2j-b# On Sat, Feb 16, 2013 at 4:18 PM, Bill whl...@gmail.com wrote: Intermittent problems are tough without the full config but have you tried making sure it is not a hardware issue? Can you match the failed calls to failed pings? Have you some debugs of your sip messages? The study list needs more information to try to help with the issue. Bill On Feb 16, 2013, at 2:25 PM, Jason Lee jas7...@gmail.com wrote: I'm running into an issue with my CUE module. I'm getting intermittent fast-busy when
[OSL | CCIE_Voice] Custom Tones
All, I have continually struggled with custom tones for a while now. I'm working on the 5LB Lab 1 today and have the preserve CBarge configuration in place. As I have it configured I'm expecting to hear one tone on entry and 2 when a call exits the call. What I'm actually hearing is 2 on join and nothing on leave. Here's the config. Can anyone see anything that I'm doing wrong? r2800-2j-b#sh run Building configuration... Current configuration : 9095 bytes ! ! Last configuration change at 17:35:03 GMT Sun Feb 17 2013 ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname r2800-2j-b ! boot-start-marker boot system flash boot-end-marker ! card type e1 0 1 card type t1 1 logging message-counter syslog enable password cisco ! no aaa new-model clock timezone GMT 0 no network-clock-participate slot 1 network-clock-participate wic 1 network-clock-select 1 E1 0/1/0 ! dot11 syslog ip source-route ! ! ip cef ip dhcp excluded-address 192.168.106.0 192.168.106.119 ip dhcp excluded-address 192.168.106.130 192.168.106.255 ! ip dhcp pool phn2 host 192.168.106.130 255.255.255.0 client-identifier 01c8.f9f9.d739.77 default-router 192.168.106.1 option 150 ip 192.168.100.100 192.168.100.101 ! ip dhcp pool voip network 192.168.106.0 255.255.255.0 option 150 ip 192.168.100.100 192.168.100.101 default-router 192.168.106.1 ! --More-- .Feb 17 17:35:03.037: %SYS-5-CONFIG_I: Configured from console !e no ip domain lookup no ipv6 cef ! multilink bundle-name authenticated ! ! ! ! isdn switch-type primary-net5 ! ! ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip fax protocol cisco ! ! ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 ! ! ! ! voice class h323 1 h225 timeout tcp establish 3 ! ! ! ! voice class custom-cptone leave dualtone conference frequency 300 cadence 400 400 400 ! voice class custom-cptone join dualtone conference frequency 300 cadence 400 ! ! ! ! ! ! ! ! voice translation-rule 1 rule 1 /.+\(\)$/ /\1/ ! voice translation-rule 9 rule 1 /^[0-8]/ /9\0/ ! voice translation-rule 23 rule 1 /2.../ /001202555\0/ type any international plan any isdn rule 2 /3.../ /001408387\0/ type any international plan any isdn ! voice translation-rule 97 rule 4 // // type any subscriber plan any isdn ! voice translation-rule 910 rule 4 // // type any national plan any isdn ! voice translation-rule 911 rule 4 // // type any unknown plan any unknown ! voice translation-rule 971 rule 1 /4.../ /+44207796\0/ rule 4 // // type any subscriber plan any isdn ! voice translation-rule 9011 rule 4 // // type any international plan any isdn ! voice translation-rule 9101 rule 1 /4.../ /+44207796\0/ rule 4 // // type any national plan any isdn ! voice translation-rule 9111 rule 1 /4...$/ /7796\0/ rule 4 // // type any unknown plan any unknown ! voice translation-rule 90111 rule 1 /4.../ /+44207796\0/ rule 4 // // type any international plan any isdn ! ! voice translation-profile 23 translate called 23 ! voice translation-profile 9 translate calling 1 translate called 9 ! voice translation-profile 9011 translate calling 90111 translate called 9011 ! voice translation-profile 910 translate calling 9101 translate called 910 ! voice translation-profile 911 translate calling 9111 translate called 911 ! voice translation-profile 97 translate calling 971 translate called 97 ! voice translation-profile strip translate called 1 ! ! voice-card 0 dsp services dspfarm ! ! ! ! ! archive log config hidekeys ! ! ! ! ! controller E1 0/1/0 pri-group timeslots 1-3,16 ! controller E1 0/1/1 ! controller T1 1/0 cablelength long 0db ! controller T1 1/1 cablelength long 0db ! ! ! ! ! interface Loopback0 ip address 192.168.96.2 255.255.255.255 h323-gateway voip bind srcaddr 192.168.96.2 ! interface GigabitEthernet0/0 no ip address duplex auto speed auto ! interface GigabitEthernet0/0.105 encapsulation dot1Q 105 native ip address 192.168.105.1 255.255.255.0 ! interface GigabitEthernet0/0.106 encapsulation dot1Q 106 ip address 192.168.106.1 255.255.255.0 ! interface Service-Engine0/0 ip unnumbered GigabitEthernet0/0.106 service-module ip address 192.168.106.2 255.255.255.0 service-module ip default-gateway 192.168.106.1 ! interface GigabitEthernet0/1 no ip address shutdown duplex auto speed auto ! interface FastEthernet0/3/0 shutdown ! interface FastEthernet0/3/1 shutdown ! interface FastEthernet0/3/2 shutdown ! interface FastEthernet0/3/3 shutdown ! interface Serial0/0/0 no ip address encapsulation frame-relay IETF no fair-queue frame-relay lmi-type ansi ip rsvp bandwidth ! interface Serial0/0/0.1 point-to-point description FR-WAN INTERFACE - DLCI 102 ip address 192.168.111.10 255.255.255.252 shutdown frame-relay interface-dlci 102 ip rsvp bandwidth 64 !
Re: [OSL | CCIE_Voice] SRST transfer system and pattern
I'm adding secondary dialtone to my CUCME and SRST configurations as well. In my mind, we should be trying to preserve as much of the CUCM configuration as possible. Not sure that it helps with grading, but better safe than sorry I guess. On Sun, Feb 17, 2013 at 4:58 AM, Pixar Perfect pixarperf...@live.comwrote: Thanks, makes sense. One of those few configurations on the exam that sticks to the design guidelines field deployments. :) :) -- Date: Sat, 16 Feb 2013 17:48:16 -0600 Subject: Re: [OSL | CCIE_Voice] SRST transfer system and pattern From: ramcharan.a...@gmail.com To: corygray22...@hotmail.com CC: pixarperf...@live.com; ccie_voice@onlinestudylist.com Hi, As per cisco CME design guide these commands are necessary. Please refer cisco CME SRND. http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/srnd/design/guide/clproc.html#wp1068396 Regards, Ramcharan Arya CCIE # 28926 ( RS) On Fri, Feb 15, 2013 at 4:51 PM, Cory Gray corygray22...@hotmail.comwrote: I have had several conversations with people on this. Everyone can easily make SRST work but scoring points seems to be the trickiest thing in the lab. So I do not think anyone knows for sure what should or should not be on the “template” I have never scored any points there so I cannot give an OPINION on what should or should not be there. People say they score points and then go with the same template on the next lab and get 0 so it is a mystery. People can share templates without breaking NDA since the question is never discussed. Getting the question right is the easy part! ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Pixar Perfect *Sent:* Friday, February 15, 2013 5:26 PM *To:* CCIE Voice OSL *Subject:* [OSL | CCIE_Voice] SRST transfer system and pattern ** ** transfer-system full-consultdo we need to specify this? I thought by default it is wnabled but I read on voiceie forum someone scored nothing on SRST adn the only conclusion was the transfersystem consult was missing. Any thoughts? ** ** srst mode auto-provision all srst ephone description SRST-EPHONES-CME srst dn template 1 srst dn line-mode octo max-ephones 10 max-dn 10 preference 2 no-reg both ip source-address 10.10.1.13 SiteC Loopback port 2000 time-zone 42 max-conferences 8 gain -6 call-forward pattern .T time-webedit * transfer-system full-consult* * transfer-pattern .T* ___ 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] Custom Tones
I'll give it a go tomorrow. I already reverted my pod this evening. I'll be doing another lab tomorrow, so I should be able to test this put by tomorrow afternoon. Sent from my iPad On Feb 17, 2013, at 9:14 PM, Bill whl...@gmail.com wrote: I think Justin might be on to it but it has been a while since I have done this in the lab. Sent from my iPad On Feb 17, 2013, at 3:06 PM, Justin Carney justin.s.car...@gmail.com wrote: I haven't tested this recently, but it may help to make the join/leave tones use different frequencies, as well as using different time intervals for the cadence. I'm not sure why you're getting these strange results (two tones on join when your cadence only shows one and no tone on leave), but there may be some strange feature (or bug) that has to do with both join and leave using the same frequency. voice class custom-cptone leave dualtone conference frequency 300 cadence 400 500 600 ! voice class custom-cptone join dualtone conference frequency 700 cadence 800 -Justin On Sun, Feb 17, 2013 at 1:56 PM, William Bell b...@ucguerrilla.com wrote: I don't have an answer for you. However, I can confirm that I have noticed the same behavior. When I have associated custom tones for join/leave events, I only hear the tone on join. Nada on leave. I haven't figured it out yet. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Feb 17, 2013, at 12:39 PM, Jason Lee wrote: All, I have continually struggled with custom tones for a while now. I'm working on the 5LB Lab 1 today and have the preserve CBarge configuration in place. As I have it configured I'm expecting to hear one tone on entry and 2 when a call exits the call. What I'm actually hearing is 2 on join and nothing on leave. Here's the config. Can anyone see anything that I'm doing wrong? r2800-2j-b#sh run Building configuration... Current configuration : 9095 bytes ! ! Last configuration change at 17:35:03 GMT Sun Feb 17 2013 ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname r2800-2j-b ! boot-start-marker boot system flash boot-end-marker ! card type e1 0 1 card type t1 1 logging message-counter syslog enable password cisco ! no aaa new-model clock timezone GMT 0 no network-clock-participate slot 1 network-clock-participate wic 1 network-clock-select 1 E1 0/1/0 ! dot11 syslog ip source-route ! ! ip cef ip dhcp excluded-address 192.168.106.0 192.168.106.119 ip dhcp excluded-address 192.168.106.130 192.168.106.255 ! ip dhcp pool phn2 host 192.168.106.130 255.255.255.0 client-identifier 01c8.f9f9.d739.77 default-router 192.168.106.1 option 150 ip 192.168.100.100 192.168.100.101 ! ip dhcp pool voip network 192.168.106.0 255.255.255.0 option 150 ip 192.168.100.100 192.168.100.101 default-router 192.168.106.1 ! --More-- .Feb 17 17:35:03.037: %SYS-5-CONFIG_I: Configured from console !e no ip domain lookup no ipv6 cef ! multilink bundle-name authenticated ! ! ! ! isdn switch-type primary-net5 ! ! ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip fax protocol cisco ! ! ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 ! ! ! ! voice class h323 1 h225 timeout tcp establish 3 ! ! ! ! voice class custom-cptone leave dualtone conference frequency 300 cadence 400 400 400 ! voice class custom-cptone join dualtone conference frequency 300 cadence 400 ! ! ! ! ! ! ! ! voice translation-rule 1 rule 1 /.+\(\)$/ /\1/ ! voice translation-rule 9 rule 1 /^[0-8]/ /9\0/ ! voice translation-rule 23 rule 1 /2.../ /001202555\0/ type any international plan any isdn rule 2 /3.../ /001408387\0/ type any international plan any isdn ! voice translation-rule 97 rule 4 // // type any subscriber plan any isdn ! voice translation-rule 910 rule 4 // // type any national plan any isdn ! voice translation-rule 911 rule 4 // // type any unknown plan any unknown ! voice translation-rule 971 rule 1 /4.../ /+44207796\0/ rule 4 // // type any subscriber plan any isdn ! voice translation-rule 9011 rule 4 // // type any international plan any isdn ! voice translation-rule 9101 rule 1 /4.../ /+44207796\0/ rule 4 // // type any national plan any isdn ! voice translation-rule 9111 rule 1 /4...$/ /7796\0/ rule 4 // // type any unknown plan any unknown ! voice translation-rule 90111 rule 1 /4.../ /+44207796\0/ rule 4 // // type any international plan any isdn ! ! voice translation-profile 23 translate called 23 ! voice translation-profile 9 translate calling 1 translate called 9 ! voice translation-profile 9011 translate calling 90111 translate called 9011 ! voice translation-profile 910
Re: [OSL | CCIE_Voice] OWLE Lab 4 CME-SRST Question
Bill, I'm with you. I try to avoid number expansions to simplify things. I set up similar scenarios exactly as you have in your first solution. I also number my translation rules, patterns, and dial-peers in a manner that allows me to call them all up using the section option on show commands. That way I can look at everything involved in the call with one command... On Sat, Feb 16, 2013 at 5:50 AM, Bill whl...@gmail.com wrote: Bill there is no preference to method just results so either method will work. I prefer voice translations myself Bill On Feb 15, 2013, at 10:38 PM, William Bell b...@ucguerrilla.com wrote: In OWLE Lab 4 there is a requirement to allow 4-digit dialing to Site A and Site B from Site C, while Site C is in SRST mode. I always handle this with the following config: voice translation-rule 91051 rule 1 /^3...$/ /1408387\0/ type any international plan any isdn rule 2 /^2...$/ /1202555\0/ type any international plan any isdn voice translation-profile 91050 translate called 91051 dial-peer voice 91050 pots translation-profile outgoing 91050 destination-pattern [23]...$ port 0/3/0:15 In the solution guide, it is handled in the following manner: voice translation-profile 900 translate called 900 ! voice translation-rule 900 rule 1 // // type any international plan any isdn ! dial-peer voice 900 pots destination-pattern 9001.. port 0/0/0:15 forward-digits 11 translation-profile out 999 num-exp 2...$ 90012025552... num-exp 3...$ 90014083873... Both options achieve the desired result but I am wondering if the latter option is preferred for any technical reason. Thanks in advance, -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla ___ 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] CUE Dropped Calls
I'm running into an issue with my CUE module. I'm getting intermittent fast-busy when calling into it. At first I though it was a random thing, but right now it is dropping 50% of all calls. 2 calls will go though, 2 will fail, and so on. I see this behavior in both CME and CUCM integrations. It occurs from phones local and remote to SC. Has anyone run into anything like this before? Here's some of the relevant configuration: interface Service-Engine0/0 ip unnumbered GigabitEthernet0/0.106 service-module ip address 192.168.106.2 255.255.255.0 service-module ip default-gateway 192.168.106.1 ! ip route 192.168.106.2 255.255.255.255 Service-Engine0/0 ___ 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] [CME WEB ADMIN]
I'm going to agree with Cory here. You will be much faster working through the CLI. Time is of essence in the lab! On Thu, Feb 14, 2013 at 9:51 AM, Cory Gray corygray22...@hotmail.comwrote: It is permitted during the lab but I do not know of anyone who uses it. Some use GUI for CUE but I cannot see how it would save you time for CME. If you feel it does, go for it! ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *ie ravindra *Sent:* Thursday, February 14, 2013 9:42 AM *To:* CCIE Study *Subject:* [OSL | CCIE_Voice] [CME WEB ADMIN] ** ** Hi folks, As I know there are certain control in CUCME using web administration. I believe if we used web administration page over CME configuration part might easier. But is it permitted during the lab time. How we can aproach ? 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
Re: [OSL | CCIE_Voice] ISDN signaling config
I have never done this. Anyone else? On Wed, Feb 13, 2013 at 9:11 PM, Pixar Perfect pixarperf...@live.comwrote: Is there a need to enable(check) Setup non-ISDN Progress Indicators IE Enable on the MGCP GW page ? ___ 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] Directory Number Configuration
I typically do not configure line level CSS to keep it simple. I also only use set VM profile if I have to integrate both CUC and CUE. If that isn't a requirement, I just set the profile to Default. On Wed, Feb 13, 2013 at 7:44 PM, Ben John benjoh...@hotmail.com wrote: Directory Number Setting it is a good idea to configure VM profile and CSS ? Want to avoid losing points. Ben ___ 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] Tab on the LAB exam
For sure. If they don't specify, I usually go with unsolicited. On Wed, Jan 9, 2013 at 9:46 PM, Bill Lake whl...@gmail.com wrote: the kind they tell you to use On Wed, Jan 9, 2013 at 8:15 PM, CCIEing aboaz...@gmail.com wrote: So, What type of MWI is better to use ?? outcalling , unsolicited , or sub-notify ?? On Thu, Jan 10, 2013 at 3:58 AM, CCIEing aboaz...@gmail.com wrote: Oh thanks a lot guys .. On Thu, Jan 10, 2013 at 3:31 AM, Gurpreet Singh Kukreja tycoononway1...@gmail.com wrote: Yup On Wed, Jan 9, 2013 at 6:26 PM, CCIEing aboaz...@gmail.com wrote: Hello Friends, I have a small question about the exam, does the Tab (to complete commands ) in the routers CLI is enabled ? Thanks ___ 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 ___ 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] Unity connection recording for UCCX prompts
I think it's best if you know multiple ways to record prompts. There's no telling what you might run into in the lab. I wouldn't take it for granted that any one method would be available. All methods rely on some form of integration. If for whatever reason you can't get the integration to work you may need an alternate method. On Fri, Jan 4, 2013 at 11:00 AM, Cory Gray corygray22...@hotmail.comwrote: For CUC, I would use Greetings Administrator ** ** For CUCCX I would use the recording script. It takes 2 seconds to make.** ** ** ** Because they are Call in and record methods, they are guaranteed to work. I cannot imagine the lab telling you how to record your prompt. I would think either it would be provided for you or you would have to record it for added complexity. I seriously doubt they would go further and tell you what method to use. ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Derek Wyss *Sent:* Friday, January 04, 2013 10:10 AM *To:* William Bell *Cc:* ccie_voice@onlinestudylist.com Voice; singh *Subject:* Re: [OSL | CCIE_Voice] Unity connection recording for UCCX prompts ** ** Bill, I haven't personally seen a scenario with the recording script not working. Unless they specifically ask for 1 way or the other. Derek On Fri, Jan 4, 2013 at 9:06 AM, William Bell b...@ucguerrilla.com wrote: Derek, ** ** Is it possible to expand on your statement without violating NDA? I ask because I struggle trying to imagine a scenario where I could get to UCCX to run the script that plays the prompts but I would be unable to create a script that records the prompts (thus forcing me to use CUC or some other method). ** ** -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla ** ** ** ** ** ** On Jan 4, 2013, at 7:58 AM, Derek Wyss wrote: I would recommend knowing how to do it both ways as certain circumstances might require it. Derek On Thu, Jan 3, 2013 at 11:21 PM, William Bell b...@ucguerrilla.com wrote: I assume everyone has their own approach here. I do the following: ** ** 1. For Unity Connection recordings (call handlers) I use CUGA ** ** 2. For UCCX prompts, I write a script in UCCX and record/upload the prompts from the UCCX server ** ** 3. For BACD prompts, I use the UCCX to record the prompt / upload to UCM-TFTP / TFTP copy the file to flash ** ** 4. For CUE prompts, I use the CUE prompt management app ** ** -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla ** ** ** ** ** ** On Jan 4, 2013, at 12:02 AM, singh wrote: ** ** HI Guys, I am planning to use Unity connection to record and download prompts for the UCCX scripts . I am just wondering if this is the best approach or a recording script needs to be written on UCCX. Also from machine on which UCCX is installed can the Unity connection web interface be accessed directly ? -singh Get Yourself a cool, short *@in.com* Email ID now!http://www3.in.com/sso/commonregister.php?ref=INutm_source=inviteutm_medium=outgoing ___ 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 ___ 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] Call-Forward to VM
All, Having a weird problem. I have CUC integrated with CUCM via SCCP. I'm able to access the CUC server by dialing the VM pilot or pressing the messages button on the phone. When I forward calls to VM under line configuration using the VM checkbox I get a fast-busy. If I uncheck the box and manually enter the VM pilot number it works fine. Has anyone ever run into this problem? thanks, Jason ___ 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] SIP CUCME Registration Failure - TFTP File Not Present
All, I'm running into and issue trying to register SIP phones to my CUCME rotuer. I'm using SIP Digest Authentication. For some reason the SIP00070EA62AA9.cnf file is not being created in flash and my phone isn't getting its configuration. I've attached the configs and debug output. I've tried removing all the SIP CUCME configuration, reloading the router, adding and removing the phone, and a ton of create profile commands with no luck. Any ideas? ip dhcp excluded-address 192.168.106.1 192.168.106.10 ! ip dhcp pool VoIP network 192.168.106.0 255.255.255.0 default-router 192.168.106.1 option 150 ip 192.168.106.1 ! voice service voip fax protocol cisco sip bind control source-interface GigabitEthernet0/0.106 bind media source-interface GigabitEthernet0/0.106 registrar server ! voice register global mode cme source-address 192.168.106.1 port 5060 max-dn 10 max-pool 5 authenticate register tftp-path flash: create profile sync 144234935113 ! voice register dn 1 number 3002 name Br2 Phone 2 ! voice register dialplan 1 type 7940-7960-others pattern 1 3... ! voice register pool 1 id mac 0007.0EA6.2AA9 number 1 dn 1 dtmf-relay rtp-nte username User1 password cisco codec g711ulaw DEBUG OUTPUT (debug tftp events): Aug 3 21:38:39.653: TFTP: Looking for CTLSEP00070EA62AA9.tlv Aug 3 21:38:39.677: TFTP: Looking for SEP00070EA62AA9.cnf.xml Aug 3 21:38:39.697: TFTP: Looking for SIP00070EA62AA9.cnf Aug 3 21:38:39.717: TFTP: Looking for MGC00070EA62AA9.cnf Aug 3 21:38:39.749: TFTP: Looking for XMLDefault.cnf.xml Aug 3 21:38:39.749: TFTP: Opened system:/its/vrf1/XMLDefault.cnf.xml, fd 7, size 2813 for process 194 Aug 3 21:38:39.757: TFTP: Finished system:/its/vrf1/XMLDefault.cnf.xml, time 00:00:00 for process 194 Aug 3 21:38:48.545: TFTP: Looking for SIPDefault.cnf Aug 3 21:38:48.549: TFTP: Opened flash:/SIPDefault.cnf, fd 7, size 1948 for process 194 Aug 3 21:38:48.561: TFTP: Finished flash:/SIPDefault.cnf, time 00:00:00 for process 194 Aug 3 21:38:48.753: TFTP: Looking for SIP00070EA62AA9.cnf Aug 3 21:39:20.220: TFTP: Looking for SIP00070EA62AA9.cnf Aug 3 21:39:51.254: TFTP: Looking for SIP00070EA62AA9.cnf It just keeps looking for the file over and over. Consequently, it doesn't get the username and password so it returns a constant 401 Unauthorized when using the debug ccsip messages. Thanks, Jason ___ 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] SIP CUCME Registration Failure - TFTP File Not Present
You all nailed it! Can't believe I missed that... Thanks for the assist. Maybe now my headache will go away! On Wed, Aug 3, 2011 at 6:13 PM, Kshitij Singhi martinian.ksin...@gmail.comwrote: Hi Jason, Add the type command under the voice register pool, do a create profile under voice register global and then test. On Thu, Aug 4, 2011 at 3:15 AM, ccie_voice-requ...@onlinestudylist.comwrote: 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. Re: real world QOS (Abel ...) 2. SIP CUCME Registration Failure - TFTP File NotPresent (Jason Lee) 3. Re: SIP CUCME Registration Failure - TFTP FileNot Present (Abel ...) -- Message: 1 Date: Wed, 3 Aug 2011 16:56:37 -0400 From: Abel ... midga...@gmail.com To: Bill Lake whl...@gmail.com Cc: OSL Questions ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] real world QOS Message-ID: CAJ6s= mayernqqp+m1s3ca5mnwj9zs1dcvtac_nykgstnu4v...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Skip the NBAR Part, I see is only ports and IP in the configurations. On Wed, Aug 3, 2011 at 4:43 PM, Abel ... midga...@gmail.com wrote: Try to undo the AutoQOS command with the script. Then paste or write yourself the commands create by AutoQoS, if problem persist try to configure QoS without NBAR. This is just to check if is an IOS issue, Also I see you're using IPS 15.0, try it with IOS 12.4T on lasted versions. Regards On Wed, Aug 3, 2011 at 2:04 PM, Bill Lake whl...@gmail.com wrote: Tried cut and paste from Serial 0/1/0 and same error was seen I am wondering if it is just not a bad IOS load, bug or limitation of the 1941 as it won't even let us run auto qos voip on the port which should do this for us. On Wed, Aug 3, 2011 at 1:56 PM, Michael Miller kf4...@gmail.com wrote: Just a first observation: The policy map is defined as policy-map AutoQoS-Policy-UnTrust The error says policy map AutoQos-Policy-Untrust not configured. I believe that policy map names are case sensitive. Are you using the proper case when applying service policy to the interface? Thanks, Mike On Wed, Aug 3, 2011 at 6:50 PM, Bill Lake whl...@gmail.com wrote: Hello everyone, I have been helping a customer try to resolve a QOS issue. They have a HQ and 2 branch locations. These are connected by a data T1 connection. They are using all Cisco 1941 routers and we have QOS working perfectly on one circuit but the other always errors if we try auto qos or manually adding it. Gives the error *% policy map AutoQos-Policy-Untrust not configured.* I have tried to reproduce it with my lab but since I have all 2800 series, I can not. I do not know if it is a IOS limitation or a router limitation. Anyone with any insight into this would be greatly appriciated. Here is the configuration with all crypto and passwords removed. version 15.0 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! boot-start-marker boot system flash:c1900-universalk9-mz.SPA.150-1.M5.bin boot-end-marker ! logging buffered 51200 warnings ! no aaa new-model ! no ipv6 cef ip source-route ip cef ! ! ! ! ip domain name yourdomain.com multilink bundle-name authenticated ! ! ! ! ! ! ! class-map match-any AutoQoS-VoIP-Remark match ip dscp ef match ip dscp cs3 match ip dscp af31 class-map match-any AutoQoS-VoIP-Control-UnTrust match access-group name AutoQoS-VoIP-Control class-map match-any AutoQoS-VoIP-RTP-UnTrust match protocol rtp audio match access-group name AutoQoS-VoIP-RTCP ! ! policy-map AutoQoS-Policy-UnTrust class AutoQoS-VoIP-RTP-UnTrust priority percent 70 set dscp ef class AutoQoS-VoIP-Control-UnTrust bandwidth percent 5 set dscp af31 class AutoQoS-VoIP-Remark set dscp default class class-default fair-queue ! ! ! ! ! interface GigabitEthernet0/0 ip address 10.10.10.1 255.255.255.248 duplex auto speed auto ! interface GigabitEthernet0/1 description LAN ip address 192.168.0.150 255.255.255.0 duplex auto speed auto ! interface Serial0/0/0 description BR1_P2P ip address 172.16.156.5 255.255.255.252 encapsulation ppp fair-queue service-module t1 timeslots 1-24
Re: [OSL | CCIE_Voice] CCIE VOICE LAB PHONES [7940-7960] CME SIP QUESTION.
I'm following the strategy of everyone else and using CUCM to dictate the version of SIP running on the phones in my lab. I have 7960s running the default version of SIP software that is available on CUCM 7.0(1). It shows up as P0S3-08-8-00 in the device defaults page. I have registered phones running that version of software many times to both CUCM and CUCME without issues. Well except for the normal growing pain issues associated with registering SIP phones to CUCME. HTH, Jason On Thu, Jul 21, 2011 at 8:22 AM, Bill Lake whl...@gmail.com wrote: Use your CUCM to load sip, even in the lab or real world it works best. Then once upgraded point the tftp (option 150) back at the CME and you then can get registered on the CME. I have done this several times without issue. However, I seldom work with 7960, mostly 61/62/65. I think from reading the forums that the 7960's might be more difficult to get SIP working on but I know getting SIP working with CME is very difficult (which is why I have VM of CUCM on my PC, to upgrade phones in the field that need SIP and not to local CME or over slow WAN to CUCM) On Thu, Jul 21, 2011 at 12:10 AM, michael.se...@compucom.com wrote: I’m working on putting together my home lab and have started working on CME. ** ** I have researched the 7940 and 7960 and on CUCM7 they are supported with the following SIP images 7940 SIP-*P0S3-8-12-00* and 7960 SIP-* SIP41.9-0-2SR1S*. Will these same images work on CME. Does someone out there know what SIP image to download for CME to support these phones for SIP? Any suggestions would be appreciated. ** ** ** ** ___ 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] Volume 2 Lab 1 - 4.3 SIP Calling Back to CUCM
Hi All, I'm running through V2L1 on section 4.3. I'm running into an issue that didn't seem to get Vikram. When calling from my Br2 SIP phone back to a CUCM phone I get ring and when I answer it hangs up, but the Br2 SIP phone keeps ringing. In the 4.2 section we had to uncheck the Wait for Far End H.245 Terminal Capability Set tick box and that prevented it from happening when the CUBE was involved. In this section the CUBE is not a member of the call but I'm seeing the same symptom. I've included some of the relevant configuration. My IPs are different though. Any ideas? HQ RTR: gatekeeper zone local UCM cisco.com 192.168.96.10 zone local UCME cisco.com outvia VGK zone local VGK cisco.com zone prefix UCM 1... gw-priority 10 gk-trunk2 zone prefix UCM 1... gw-priority 9 gk-trunk1 zone prefix UCME 3... zone prefix UCM 5... gw-priority 10 gk-trunk2 zone prefix UCM 5... gw-priority 9 gk-trunk1 gw-type-prefix 1#* default-technology no shutdown ! dial-peer voice 1 pots incoming called-number . direct-inward-dial ! dial-peer voice 5000 voip destination-pattern 5... voice-class codec 1 voice-class h323 1 session target ipv4:192.168.100.101 dtmf-relay h245-signal h245-alphanumeric no vad ! dial-peer voice 5001 voip preference 1 destination-pattern 5... voice-class h323 1 session target ipv4:192.168.100.100 dtmf-relay h245-alphanumeric no vad ! dial-peer voice 3000 voip incoming called-number 3... dtmf-relay h245-alphanumeric no vad ! dial-peer voice 3001 voip destination-pattern 3... session target ras dtmf-relay h245-alphanumeric codec g711ulaw no vad Br2 Router: dial-peer voice 5000 voip destination-pattern 5... session target ras dtmf-relay h245-alphanumeric no vad ! dial-peer voice 1000 voip destination-pattern 1... session target ras dtmf-relay h245-alphanumeric no vad Thanks, Jason ___ 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
