Re: [OSL | CCIE_Voice] Base configs and Dial Plan mapping
+1 for that. Awesome information. Imagine this come out in one of the next revision of the exam on backup Route List and on purpose remove the ability to change this parameter on Call Manager. (4 points)!!! Thank you very much for sharing. Regards, --Somphol. On Sat, Oct 19, 2013 at 3:10 AM, Bill Hatcher wrote: > That's great information Bill. I think I might start leveraging that > command on my real world deployments. > > Sent from my iPhone so please excuse any spelling mistakes. > > Bill Hatcher > > On Oct 18, 2013, at 9:42 AM, William Bell wrote: > > Bill, > > You can read about the command here: > http://www.cisco.com/en/US/docs/ios/12_3t/voice/command/reference/vrht_d1_ps5207_TSD_Products_Command_Reference_Chapter.html#wp1459139 > > The important bit is: > When the *dial-peer outbound status-check pots *command is configured, if > the voice-port configured under an outbound POTS dial-peer is down, that > dial-peer is excluded while matching the corresponding destination-pattern. > Therefore, if there are no other matching outbound POTS dial-peers for the > specified destination-pattern, the gateway will disconnect the call with a > cause code of 1 (Unallocated/unassigned number), > > So, when you have this command enabled (default) AND you have a single PRI > AND that PRI is down, call set up request from UCM to the VG will result in > a response of unallocated/unassigned. Why? Because we have told the router > to monitor the status of the PRI and "intelligently" detect when it is > down. When it is down, the dial-peer is no longer evaluated during call > setup. > > By turning this off, we are basically telling the VG to go ahead and try > to use the busted PRI. Which then results in a different kind of setup > error that will let the CUCM know it should continue hunting through its > RG/RL configuration. > > Lots of people leverage the service parameter I mentioned below to route > around PRIs that are off line. That is probably fine for the purposes of > the IE lab. I prefer to disable status checking at the GW level. > > -Bill > > > > > -- > William Bell, CCIE #38914 > blog: http://ucguerrilla.com > twitter: @ucguerrilla > > > > > On Oct 18, 2013, at 10:06 AM, Bill Hatcher wrote: > > Bill, > > One other question, I'm not familiar with the command "no dial-peer out > status pots" What's it do? > > > On Fri, Oct 18, 2013 at 7:44 AM, William Bell wrote: > >> I documented my strategy in my blog if interested. Part 2 in the series >> focuses on building various tables and the read-through portion of the exam: >> >> http://ucguerrilla.com/2013/04/ccie-voice-lab-strategy-part-2-read.html >> >> Looking back at my notes, I have the following Ent / Service params that >> I updated by default: >> >> >> *Enterprise Parameters:* >> * >> * >> >>- Auto Registration Protocol: SCCP >>- BLF for Call Lists: Enabled >>- Advertise G722 Codec: Disabled >>- URL Authentication: set IP instead of name >>- URL Directories: set IP instead of name >>- URL information: set IP instead of name >>- URL Services: set IP instead of name >>- Connection Monitor Duration: 60 (or do this at a device pool level) >> >> >> *Service Parameters* >> >>- BRQ Enabled: True >>- T302 timer: 5000 >>- H225 T302 timer: 5000 >>- G722 codec enabled: Disabled >>- iLBC codec enabled: Disabled >>- Intraregion Audio codec default: G729 >>- Inter-region Audio codec default: G729 >>- Automated Alternate Routing: True >>- Enable Mobile Voice Access: True >>- Inbound Calling Search Space for Remote Destination: Remote >>Destination Profile + Line Calling Search Space >>- System Remote Access Block Numbers: >>- Transfer on-hook enabled: True >>- Display Original Calling Number on Transfer from Unity: True >>- Max Forward unregistered hops to DN: 1 >>- Allow peer to preserve h323 calls: True/*need to add >>appropriate configuration on h323*/ >> >> >> Another service parameter I have seen people modify is the "stop routing >> on unallocated number" parameter. People mod this to allow calls to hunt >> around a H323 gateway that has a PRI which is down. I didn't use this >> method because I think it is the wrong approach to fixing that problem. I >> leveraged the IOS config command: "no dial-peer out status pots" >> >> >> HTH. >> >> -Bill >> >> -- >> William Bell, CCIE #38914 >> blog: http://ucguerrilla.com >> twitter: @ucguerrilla >> >> >> >> >> On Oct 17, 2013, at 1:39 PM, Bill Hatcher wrote: >> >> My test is just a couple of weeks away, and I've been reading different >> blogs on how to maximize your time. The one thing I'm really struggling >> with is mapping out my dial-plan during my read through of the lab. I >> would love to hear what others are doing. >> >> I have also been building base router configs for h323, gatekeeper, mgcp, >> srst,sip, etc so that I can practice quickly configuring those on the >> routers. >> >> One of the thi
Re: [OSL | CCIE_Voice] Base configs and Dial Plan mapping
Hi Bill, I have adjusted it a little bit more to reduce the tab realignment while editing. It turns out that this creates more space where I can put in extra annotation such as route pattern. I think creating it from scratch could be done within 3 minutes. See a bit of a video clip here - http://www.youtube.com/watch?v=Cl9nANVgbms The text file can also available here - https://www.dropbox.com/s/t9bb2yo6x0wc66g/Dial-Plan-on-Notepad-Demo.TXT Regards, --Somphol. On Sat, Oct 19, 2013 at 1:15 AM, Bill Hatcher wrote: > Great template!! I like doing it this way better than on paper. I make > to many mistakes on paper and can hardly read what I wrote. Thanks!! > > > On Fri, Oct 18, 2013 at 2:53 AM, Somphol Boonjing wrote: > >> >> The one thing I'm really struggling with is mapping out my dial-plan >> during my read through of the lab. I would love to hear what others are >> doing. >> >> In my previous attempts, I find it very hard too, because the questions >> are verbose and I could either spend too much time reading OR not able to >> encode it into the table form correctly in haste OR simply skipped and >> waste too much time re-reading it. >> >> Here is my plan for my next attempt.I think the key is to have my >> pre-fabricate table then I will create my table quickly and ONLY adjust it >> while I read the question. >> >> So, I would quickly create this template. The list is there for easy cut >> & paste. I will only complete Site A in during the lab, then I will just >> copy to SiteB & SiteC. The rest is just modification of the table. >> (Note: I find that using "TAB" make it easier to align the columns, it >> could be 3 or 4 TABs.) >> >> In essence, focus on [1] Pre-fabrication [2] Quick to reproduce as a >> template. The rest is depending on how quick you can decipher verbose >> question and re-adjust the table. >> >> Get the screenshot here is the following format is bad - >> https://www.dropbox.com/s/o1ftbh4katnhjcu/Quick-Dial-Plan-Template.jpg, >> The TXT version is here - >> https://www.dropbox.com/s/m2p2lte5ezk6q7d/Quick-Dial-Plan-Template.TXT >> >> I think Matthew Berry youtube is good too - >> http://www.youtube.com/watch?v=4mP5powuFUM.I just think it can be >> complement by the cookie-cutting approach to encode the table in Notepad >> that can be reproduced quickly. >> >> === >> The LIST >> === >> >> ISDN >> Unknown >> Subscriber >> National >> International >> Any >> >> === >> SiteA >> === >> >> Calling Called >> >> Emer 7D / Unknown / ISDN 7D / Unknown / ISDN >> Local 7D / Subscriber / ISDN 7D / Subscriber / ISDN >> LD 10D / National / ISDN 10D / National / ISDN >> Intl 1+10D / International / ISDN 011! / International / ISDN >> >> >> Another variation of the table format is too cater for TEHO scenario or >> BACKUP Gateway scenario. >> >> === >> SiteA >> === >> >> Calling Called >> >> Emer 7D / Unknown / ISDN 7D / Unknown / ISDN >> Local 7D / Subscriber / ISDN 7D / Subscriber / ISDN >> Local(Backup via SiteB) 7D / Subscriber / ISDN 7D / Subscriber / ISDN >> LD 10D / National / ISDN 10D / National / ISDN >> Intl 1+10D / International / ISDN 011! / International / ISDN >> >> Regards, >> --Somphol. >> >> >> On Fri, Oct 18, 2013 at 12:22 PM, VanBenschoten, Brian < >> brian.vanbenscho...@corebts.com> wrote: >> >>> I've found the QoS questions are very specific to test a certain area >>> of knowledge. They are not looking for what we would consider a "best >>> practice" system wide. I think we could skip setting the DSCP values in >>> CUCM. >>> >>> If you think the question calls for it you can have your class-map match >>> both AF31 and CS3 for signaling. >>> >>> >>> >>> *From:* ccie_voice-boun...@onlinestudylist.com [mailto: >>> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Bill Hatcher >>> *Sent:* Thursday, October 17, 2013 3:14 PM >>> *To:* probert...@gmail.com >>> *Cc:* ccievoice >>> *Subject:* Re: [OSL | CCIE_Voice] Base configs and Dial Plan mapping >>> >>> >>> >>> I agree to setting the service parameters to default first. I was >>> planning on doing that myself. As to c
Re: [OSL | CCIE_Voice] Base configs and Dial Plan mapping
That's great information Bill. I think I might start leveraging that command on my real world deployments. Sent from my iPhone so please excuse any spelling mistakes. Bill Hatcher > On Oct 18, 2013, at 9:42 AM, William Bell wrote: > > Bill, > > You can read about the command here: > http://www.cisco.com/en/US/docs/ios/12_3t/voice/command/reference/vrht_d1_ps5207_TSD_Products_Command_Reference_Chapter.html#wp1459139 > > The important bit is: > When the dial-peer outbound status-check pots command is configured, if the > voice-port configured under an outbound POTS dial-peer is down, that > dial-peer is excluded while matching the corresponding destination-pattern. > Therefore, if there are no other matching outbound POTS dial-peers for the > specified destination-pattern, the gateway will disconnect the call with a > cause code of 1 (Unallocated/unassigned number), > > So, when you have this command enabled (default) AND you have a single PRI > AND that PRI is down, call set up request from UCM to the VG will result in a > response of unallocated/unassigned. Why? Because we have told the router to > monitor the status of the PRI and "intelligently" detect when it is down. > When it is down, the dial-peer is no longer evaluated during call setup. > > By turning this off, we are basically telling the VG to go ahead and try to > use the busted PRI. Which then results in a different kind of setup error > that will let the CUCM know it should continue hunting through its RG/RL > configuration. > > Lots of people leverage the service parameter I mentioned below to route > around PRIs that are off line. That is probably fine for the purposes of the > IE lab. I prefer to disable status checking at the GW level. > > -Bill > > > > > -- > William Bell, CCIE #38914 > blog: http://ucguerrilla.com > twitter: @ucguerrilla > > > > >> On Oct 18, 2013, at 10:06 AM, Bill Hatcher wrote: >> >> Bill, >> >> One other question, I'm not familiar with the command "no dial-peer out >> status pots" What's it do? >> >> >>> On Fri, Oct 18, 2013 at 7:44 AM, William Bell wrote: >>> I documented my strategy in my blog if interested. Part 2 in the series >>> focuses on building various tables and the read-through portion of the exam: >>> >>> http://ucguerrilla.com/2013/04/ccie-voice-lab-strategy-part-2-read.html >>> >>> Looking back at my notes, I have the following Ent / Service params that I >>> updated by default: >>> >>> >>> Enterprise Parameters: >>> >>> Auto Registration Protocol: SCCP >>> BLF for Call Lists: Enabled >>> Advertise G722 Codec: Disabled >>> URL Authentication: set IP instead of name >>> URL Directories: set IP instead of name >>> URL information: set IP instead of name >>> URL Services: set IP instead of name >>> Connection Monitor Duration: 60 (or do this at a device pool level) >>> >>> Service Parameters >>> BRQ Enabled: True >>> T302 timer: 5000 >>> H225 T302 timer: 5000 >>> G722 codec enabled: Disabled >>> iLBC codec enabled: Disabled >>> Intraregion Audio codec default: G729 >>> Inter-region Audio codec default: G729 >>> Automated Alternate Routing: True >>> Enable Mobile Voice Access: True >>> Inbound Calling Search Space for Remote Destination: Remote Destination >>> Profile + Line Calling Search Space >>> System Remote Access Block Numbers: >>> Transfer on-hook enabled: True >>> Display Original Calling Number on Transfer from Unity: True >>> Max Forward unregistered hops to DN: 1 >>> Allow peer to preserve h323 calls: True/*need to add appropriate >>> configuration on h323*/ >>> >>> Another service parameter I have seen people modify is the "stop routing on >>> unallocated number" parameter. People mod this to allow calls to hunt >>> around a H323 gateway that has a PRI which is down. I didn't use this >>> method because I think it is the wrong approach to fixing that problem. I >>> leveraged the IOS config command: "no dial-peer out status pots" >>> >>> >>> HTH. >>> >>> -Bill >>> >>> -- >>> William Bell, CCIE #38914 >>> blog: http://ucguerrilla.com >>> twitter: @ucguerrilla >>> >>> >>> >>> On Oct 17, 2013, at 1:39 PM, Bill Hatcher wrote: My test is just a couple of weeks away, and I've been reading different blogs on how to maximize your time. The one thing I'm really struggling with is mapping out my dial-plan during my read through of the lab. I would love to hear what others are doing. I have also been building base router configs for h323, gatekeeper, mgcp, srst,sip, etc so that I can practice quickly configuring those on the routers. One of the things I haven't really been keeping track of are some of the service parameters that I should adjust out of habit. Here are a few that I can think of off the top of my head that I plan on tweaking at the start of the exam. Please feel free to add to them.
Re: [OSL | CCIE_Voice] Base configs and Dial Plan mapping
Bill, You can read about the command here: http://www.cisco.com/en/US/docs/ios/12_3t/voice/command/reference/vrht_d1_ps5207_TSD_Products_Command_Reference_Chapter.html#wp1459139 The important bit is: When the dial-peer outbound status-check pots command is configured, if the voice-port configured under an outbound POTS dial-peer is down, that dial-peer is excluded while matching the corresponding destination-pattern. Therefore, if there are no other matching outbound POTS dial-peers for the specified destination-pattern, the gateway will disconnect the call with a cause code of 1 (Unallocated/unassigned number), So, when you have this command enabled (default) AND you have a single PRI AND that PRI is down, call set up request from UCM to the VG will result in a response of unallocated/unassigned. Why? Because we have told the router to monitor the status of the PRI and "intelligently" detect when it is down. When it is down, the dial-peer is no longer evaluated during call setup. By turning this off, we are basically telling the VG to go ahead and try to use the busted PRI. Which then results in a different kind of setup error that will let the CUCM know it should continue hunting through its RG/RL configuration. Lots of people leverage the service parameter I mentioned below to route around PRIs that are off line. That is probably fine for the purposes of the IE lab. I prefer to disable status checking at the GW level. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Oct 18, 2013, at 10:06 AM, Bill Hatcher wrote: > Bill, > > One other question, I'm not familiar with the command "no dial-peer out > status pots" What's it do? > > > On Fri, Oct 18, 2013 at 7:44 AM, William Bell wrote: > I documented my strategy in my blog if interested. Part 2 in the series > focuses on building various tables and the read-through portion of the exam: > > http://ucguerrilla.com/2013/04/ccie-voice-lab-strategy-part-2-read.html > > Looking back at my notes, I have the following Ent / Service params that I > updated by default: > > > Enterprise Parameters: > > Auto Registration Protocol: SCCP > BLF for Call Lists: Enabled > Advertise G722 Codec: Disabled > URL Authentication: set IP instead of name > URL Directories: set IP instead of name > URL information: set IP instead of name > URL Services: set IP instead of name > Connection Monitor Duration: 60 (or do this at a device pool level) > > Service Parameters > BRQ Enabled: True > T302 timer: 5000 > H225 T302 timer: 5000 > G722 codec enabled: Disabled > iLBC codec enabled: Disabled > Intraregion Audio codec default: G729 > Inter-region Audio codec default: G729 > Automated Alternate Routing: True > Enable Mobile Voice Access: True > Inbound Calling Search Space for Remote Destination: Remote Destination > Profile + Line Calling Search Space > System Remote Access Block Numbers: > Transfer on-hook enabled: True > Display Original Calling Number on Transfer from Unity: True > Max Forward unregistered hops to DN: 1 > Allow peer to preserve h323 calls: True/*need to add appropriate > configuration on h323*/ > > Another service parameter I have seen people modify is the "stop routing on > unallocated number" parameter. People mod this to allow calls to hunt around > a H323 gateway that has a PRI which is down. I didn't use this method because > I think it is the wrong approach to fixing that problem. I leveraged the IOS > config command: "no dial-peer out status pots" > > > HTH. > > -Bill > > -- > William Bell, CCIE #38914 > blog: http://ucguerrilla.com > twitter: @ucguerrilla > > > > > On Oct 17, 2013, at 1:39 PM, Bill Hatcher wrote: > >> My test is just a couple of weeks away, and I've been reading different >> blogs on how to maximize your time. The one thing I'm really struggling >> with is mapping out my dial-plan during my read through of the lab. I would >> love to hear what others are doing. >> >> I have also been building base router configs for h323, gatekeeper, mgcp, >> srst,sip, etc so that I can practice quickly configuring those on the >> routers. >> >> One of the things I haven't really been keeping track of are some of the >> service parameters that I should adjust out of habit. Here are a few that I >> can think of off the top of my head that I plan on tweaking at the start of >> the exam. Please feel free to add to them. >> >> Enterprise Parameters >> DSCP for Phone Configuration - Set to AF31 >> DSCP for Cisco CallManager to Device Interface - Set to AF31 >> Change the Phone URL's to IP's >> Organization Top Level Domain >> Cluster Fully Qualified Domain Name >> >> Service Parameters - CallManager >> T302 Time - Know where it is if you need ot change interdigit timeout >> Call Classification - Offnet >> Builtin Bridge Enabled - True >> Device Name of GK-controlled Trunk That Will Use Port 1720
Re: [OSL | CCIE_Voice] Base configs and Dial Plan mapping
Great template!! I like doing it this way better than on paper. I make to many mistakes on paper and can hardly read what I wrote. Thanks!! On Fri, Oct 18, 2013 at 2:53 AM, Somphol Boonjing wrote: > >> The one thing I'm really struggling with is mapping out my dial-plan > during my read through of the lab. I would love to hear what others are > doing. > > In my previous attempts, I find it very hard too, because the questions > are verbose and I could either spend too much time reading OR not able to > encode it into the table form correctly in haste OR simply skipped and > waste too much time re-reading it. > > Here is my plan for my next attempt.I think the key is to have my > pre-fabricate table then I will create my table quickly and ONLY adjust it > while I read the question. > > So, I would quickly create this template. The list is there for easy cut > & paste. I will only complete Site A in during the lab, then I will just > copy to SiteB & SiteC. The rest is just modification of the table. > (Note: I find that using "TAB" make it easier to align the columns, it > could be 3 or 4 TABs.) > > In essence, focus on [1] Pre-fabrication [2] Quick to reproduce as a > template. The rest is depending on how quick you can decipher verbose > question and re-adjust the table. > > Get the screenshot here is the following format is bad - > https://www.dropbox.com/s/o1ftbh4katnhjcu/Quick-Dial-Plan-Template.jpg, > The TXT version is here - > https://www.dropbox.com/s/m2p2lte5ezk6q7d/Quick-Dial-Plan-Template.TXT > > I think Matthew Berry youtube is good too - > http://www.youtube.com/watch?v=4mP5powuFUM.I just think it can be > complement by the cookie-cutting approach to encode the table in Notepad > that can be reproduced quickly. > > === > The LIST > === > > ISDN > Unknown > Subscriber > National > International > Any > > === > SiteA > === > > Calling Called > > Emer 7D / Unknown / ISDN 7D / Unknown / ISDN > Local 7D / Subscriber / ISDN 7D / Subscriber / ISDN > LD 10D / National / ISDN 10D / National / ISDN > Intl 1+10D / International / ISDN 011! / International / ISDN > > > Another variation of the table format is too cater for TEHO scenario or > BACKUP Gateway scenario. > > === > SiteA > === > > Calling Called > > Emer 7D / Unknown / ISDN 7D / Unknown / ISDN > Local 7D / Subscriber / ISDN 7D / Subscriber / ISDN > Local(Backup via SiteB) 7D / Subscriber / ISDN 7D / Subscriber / ISDN > LD 10D / National / ISDN 10D / National / ISDN > Intl 1+10D / International / ISDN 011! / International / ISDN > > Regards, > --Somphol. > > > On Fri, Oct 18, 2013 at 12:22 PM, VanBenschoten, Brian < > brian.vanbenscho...@corebts.com> wrote: > >> I've found the QoS questions are very specific to test a certain area >> of knowledge. They are not looking for what we would consider a "best >> practice" system wide. I think we could skip setting the DSCP values in >> CUCM. >> >> If you think the question calls for it you can have your class-map match >> both AF31 and CS3 for signaling. >> >> >> >> *From:* ccie_voice-boun...@onlinestudylist.com [mailto: >> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Bill Hatcher >> *Sent:* Thursday, October 17, 2013 3:14 PM >> *To:* probert...@gmail.com >> *Cc:* ccievoice >> *Subject:* Re: [OSL | CCIE_Voice] Base configs and Dial Plan mapping >> >> >> >> I agree to setting the service parameters to default first. I was >> planning on doing that myself. As to changing the DSCP values, it all >> depends on what they ask for in the QoS section of the test is all. >> >> >> >> On Thu, Oct 17, 2013 at 2:24 PM, probert...@gmail.com < >> probert...@gmail.com> wrote: >> >> Hi, >> >> I think my strategy will be to set all Service Parameters to default >> before making changes. This way I can avoid and undesirable presets. >> >> Let me know your thoughts on this. >> >> Also why are you setting DSCP for Phone Configuration and DSCP for Cisco >> CallManager to Device Interface to AF31? Default CS3 should be good, let me >> know if I'm wrong on this? >> >> >> >> Rob >> >> >> >> On Thu, Oct 17, 2013 at 1:39 PM, Bill Hatcher >> wrote: >> >> My test is just a couple of weeks away, and I've been reading >> different blogs
Re: [OSL | CCIE_Voice] Base configs and Dial Plan mapping
Thank you all for sharing your ideas and insights. All of this is great information and I will incorporate it into my plan of attack for my Oct 29th attempt. Bill On Fri, Oct 18, 2013 at 9:06 AM, Bill Hatcher wrote: > Bill, > > One other question, I'm not familiar with the command "no dial-peer out > status pots" What's it do? > > > On Fri, Oct 18, 2013 at 7:44 AM, William Bell wrote: > >> I documented my strategy in my blog if interested. Part 2 in the series >> focuses on building various tables and the read-through portion of the exam: >> >> http://ucguerrilla.com/2013/04/ccie-voice-lab-strategy-part-2-read.html >> >> Looking back at my notes, I have the following Ent / Service params that >> I updated by default: >> >> >> *Enterprise Parameters:* >> * >> * >> >>- Auto Registration Protocol: SCCP >>- BLF for Call Lists: Enabled >>- Advertise G722 Codec: Disabled >>- URL Authentication: set IP instead of name >>- URL Directories: set IP instead of name >>- URL information: set IP instead of name >>- URL Services: set IP instead of name >>- Connection Monitor Duration: 60 (or do this at a device pool level) >> >> >> *Service Parameters* >> >>- BRQ Enabled: True >>- T302 timer: 5000 >>- H225 T302 timer: 5000 >>- G722 codec enabled: Disabled >>- iLBC codec enabled: Disabled >>- Intraregion Audio codec default: G729 >>- Inter-region Audio codec default: G729 >>- Automated Alternate Routing: True >>- Enable Mobile Voice Access: True >>- Inbound Calling Search Space for Remote Destination: Remote >>Destination Profile + Line Calling Search Space >>- System Remote Access Block Numbers: >>- Transfer on-hook enabled: True >>- Display Original Calling Number on Transfer from Unity: True >>- Max Forward unregistered hops to DN: 1 >>- Allow peer to preserve h323 calls: True/*need to add >>appropriate configuration on h323*/ >> >> >> Another service parameter I have seen people modify is the "stop routing >> on unallocated number" parameter. People mod this to allow calls to hunt >> around a H323 gateway that has a PRI which is down. I didn't use this >> method because I think it is the wrong approach to fixing that problem. I >> leveraged the IOS config command: "no dial-peer out status pots" >> >> >> HTH. >> >> -Bill >> >> -- >> William Bell, CCIE #38914 >> blog: http://ucguerrilla.com >> twitter: @ucguerrilla >> >> >> >> >> On Oct 17, 2013, at 1:39 PM, Bill Hatcher wrote: >> >> My test is just a couple of weeks away, and I've been reading different >> blogs on how to maximize your time. The one thing I'm really struggling >> with is mapping out my dial-plan during my read through of the lab. I >> would love to hear what others are doing. >> >> I have also been building base router configs for h323, gatekeeper, mgcp, >> srst,sip, etc so that I can practice quickly configuring those on the >> routers. >> >> One of the things I haven't really been keeping track of are some of the >> service parameters that I should adjust out of habit. Here are a few that I >> can think of off the top of my head that I plan on tweaking at the start of >> the exam. Please feel free to add to them. >> >> Enterprise Parameters >> DSCP for Phone Configuration - Set to AF31 >> DSCP for Cisco CallManager to Device Interface - Set to AF31 >> Change the Phone URL's to IP's >> Organization Top Level Domain >> Cluster Fully Qualified Domain Name >> >> Service Parameters - CallManager >> T302 Time - Know where it is if you need ot change interdigit timeout >> Call Classification - Offnet >> Builtin Bridge Enabled - True >> Device Name of GK-controlled Trunk That Will Use Port 1720 - Set as >> needed. >> Transfer On-hook Enabled - True (Also a great thing to do in production >> when migrating users from other phone systems) >> Block offnet to offnet transfers - Know where it's at. >> Auto Call Pickup Enabled - True >> Call Back Enabled Flag - True (Verify) >> Single Button Barge/CBarge Policy - Set to Barge unless otherwise >> directed. >> Stop routing on Unallocated Number Flag - False - H323 redundancy >> Preferred G.711 Millisecond Packet Size - 20 (Verify) >> Preferred G.729 Millisecond Packet Size - 20 (Verify) >> G722 Codec Enabled - Disabled (Unless otherwise directed) >> Intraregion Audio Codec Default - G711/G722 (Verify) >> Interregion Audio Codec Default - G729 (Verify) >> Automated Alternate Routing Enabled - True (This one gets me every time >> on AAR so I turn it on by default now) >> Enable Mobile Voice Access - Set as required >> Mobile Voice Access Number - Set as required >> System Remote Access Blocked Numbers - Set as required >> >> Service Parameters -Cisco IP Voice Media Streaming App >> Supported MOH Codecs - G711 mulaw and G729 Annex A >> >> HTH >> >> Bill Hatcher >> ___ >> For more information regarding industry leading CCIE Lab training, please >> visit www.ipexp
Re: [OSL | CCIE_Voice] Base configs and Dial Plan mapping
Bill, One other question, I'm not familiar with the command "no dial-peer out status pots" What's it do? On Fri, Oct 18, 2013 at 7:44 AM, William Bell wrote: > I documented my strategy in my blog if interested. Part 2 in the series > focuses on building various tables and the read-through portion of the exam: > > http://ucguerrilla.com/2013/04/ccie-voice-lab-strategy-part-2-read.html > > Looking back at my notes, I have the following Ent / Service params that I > updated by default: > > > *Enterprise Parameters:* > * > * > >- Auto Registration Protocol: SCCP >- BLF for Call Lists: Enabled >- Advertise G722 Codec: Disabled >- URL Authentication: set IP instead of name >- URL Directories: set IP instead of name >- URL information: set IP instead of name >- URL Services: set IP instead of name >- Connection Monitor Duration: 60 (or do this at a device pool level) > > > *Service Parameters* > >- BRQ Enabled: True >- T302 timer: 5000 >- H225 T302 timer: 5000 >- G722 codec enabled: Disabled >- iLBC codec enabled: Disabled >- Intraregion Audio codec default: G729 >- Inter-region Audio codec default: G729 >- Automated Alternate Routing: True >- Enable Mobile Voice Access: True >- Inbound Calling Search Space for Remote Destination: Remote >Destination Profile + Line Calling Search Space >- System Remote Access Block Numbers: >- Transfer on-hook enabled: True >- Display Original Calling Number on Transfer from Unity: True >- Max Forward unregistered hops to DN: 1 >- Allow peer to preserve h323 calls: True/*need to add appropriate >configuration on h323*/ > > > Another service parameter I have seen people modify is the "stop routing > on unallocated number" parameter. People mod this to allow calls to hunt > around a H323 gateway that has a PRI which is down. I didn't use this > method because I think it is the wrong approach to fixing that problem. I > leveraged the IOS config command: "no dial-peer out status pots" > > > HTH. > > -Bill > > -- > William Bell, CCIE #38914 > blog: http://ucguerrilla.com > twitter: @ucguerrilla > > > > > On Oct 17, 2013, at 1:39 PM, Bill Hatcher wrote: > > My test is just a couple of weeks away, and I've been reading different > blogs on how to maximize your time. The one thing I'm really struggling > with is mapping out my dial-plan during my read through of the lab. I > would love to hear what others are doing. > > I have also been building base router configs for h323, gatekeeper, mgcp, > srst,sip, etc so that I can practice quickly configuring those on the > routers. > > One of the things I haven't really been keeping track of are some of the > service parameters that I should adjust out of habit. Here are a few that I > can think of off the top of my head that I plan on tweaking at the start of > the exam. Please feel free to add to them. > > Enterprise Parameters > DSCP for Phone Configuration - Set to AF31 > DSCP for Cisco CallManager to Device Interface - Set to AF31 > Change the Phone URL's to IP's > Organization Top Level Domain > Cluster Fully Qualified Domain Name > > Service Parameters - CallManager > T302 Time - Know where it is if you need ot change interdigit timeout > Call Classification - Offnet > Builtin Bridge Enabled - True > Device Name of GK-controlled Trunk That Will Use Port 1720 - Set as needed. > Transfer On-hook Enabled - True (Also a great thing to do in production > when migrating users from other phone systems) > Block offnet to offnet transfers - Know where it's at. > Auto Call Pickup Enabled - True > Call Back Enabled Flag - True (Verify) > Single Button Barge/CBarge Policy - Set to Barge unless otherwise directed. > Stop routing on Unallocated Number Flag - False - H323 redundancy > Preferred G.711 Millisecond Packet Size - 20 (Verify) > Preferred G.729 Millisecond Packet Size - 20 (Verify) > G722 Codec Enabled - Disabled (Unless otherwise directed) > Intraregion Audio Codec Default - G711/G722 (Verify) > Interregion Audio Codec Default - G729 (Verify) > Automated Alternate Routing Enabled - True (This one gets me every time on > AAR so I turn it on by default now) > Enable Mobile Voice Access - Set as required > Mobile Voice Access Number - Set as required > System Remote Access Blocked Numbers - Set as required > > Service Parameters -Cisco IP Voice Media Streaming App > Supported MOH Codecs - G711 mulaw and G729 Annex A > > HTH > > Bill Hatcher > ___ > 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] Base configs and Dial Plan mapping
I documented my strategy in my blog if interested. Part 2 in the series focuses on building various tables and the read-through portion of the exam: http://ucguerrilla.com/2013/04/ccie-voice-lab-strategy-part-2-read.html Looking back at my notes, I have the following Ent / Service params that I updated by default: Enterprise Parameters: Auto Registration Protocol: SCCP BLF for Call Lists: Enabled Advertise G722 Codec: Disabled URL Authentication: set IP instead of name URL Directories: set IP instead of name URL information: set IP instead of name URL Services: set IP instead of name Connection Monitor Duration: 60 (or do this at a device pool level) Service Parameters BRQ Enabled: True T302 timer: 5000 H225 T302 timer: 5000 G722 codec enabled: Disabled iLBC codec enabled: Disabled Intraregion Audio codec default: G729 Inter-region Audio codec default: G729 Automated Alternate Routing: True Enable Mobile Voice Access: True Inbound Calling Search Space for Remote Destination: Remote Destination Profile + Line Calling Search Space System Remote Access Block Numbers: Transfer on-hook enabled: True Display Original Calling Number on Transfer from Unity: True Max Forward unregistered hops to DN: 1 Allow peer to preserve h323 calls: True/*need to add appropriate configuration on h323*/ Another service parameter I have seen people modify is the "stop routing on unallocated number" parameter. People mod this to allow calls to hunt around a H323 gateway that has a PRI which is down. I didn't use this method because I think it is the wrong approach to fixing that problem. I leveraged the IOS config command: "no dial-peer out status pots" HTH. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Oct 17, 2013, at 1:39 PM, Bill Hatcher wrote: > My test is just a couple of weeks away, and I've been reading different blogs > on how to maximize your time. The one thing I'm really struggling with is > mapping out my dial-plan during my read through of the lab. I would love to > hear what others are doing. > > I have also been building base router configs for h323, gatekeeper, mgcp, > srst,sip, etc so that I can practice quickly configuring those on the routers. > > One of the things I haven't really been keeping track of are some of the > service parameters that I should adjust out of habit. Here are a few that I > can think of off the top of my head that I plan on tweaking at the start of > the exam. Please feel free to add to them. > > Enterprise Parameters > DSCP for Phone Configuration - Set to AF31 > DSCP for Cisco CallManager to Device Interface - Set to AF31 > Change the Phone URL's to IP's > Organization Top Level Domain > Cluster Fully Qualified Domain Name > > Service Parameters - CallManager > T302 Time - Know where it is if you need ot change interdigit timeout > Call Classification - Offnet > Builtin Bridge Enabled - True > Device Name of GK-controlled Trunk That Will Use Port 1720 - Set as needed. > Transfer On-hook Enabled - True (Also a great thing to do in production when > migrating users from other phone systems) > Block offnet to offnet transfers - Know where it's at. > Auto Call Pickup Enabled - True > Call Back Enabled Flag - True (Verify) > Single Button Barge/CBarge Policy - Set to Barge unless otherwise directed. > Stop routing on Unallocated Number Flag - False - H323 redundancy > Preferred G.711 Millisecond Packet Size - 20 (Verify) > Preferred G.729 Millisecond Packet Size - 20 (Verify) > G722 Codec Enabled - Disabled (Unless otherwise directed) > Intraregion Audio Codec Default - G711/G722 (Verify) > Interregion Audio Codec Default - G729 (Verify) > Automated Alternate Routing Enabled - True (This one gets me every time on > AAR so I turn it on by default now) > Enable Mobile Voice Access - Set as required > Mobile Voice Access Number - Set as required > System Remote Access Blocked Numbers - Set as required > > Service Parameters -Cisco IP Voice Media Streaming App > Supported MOH Codecs - G711 mulaw and G729 Annex A > > HTH > > Bill Hatcher > ___ > 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] Base configs and Dial Plan mapping
>> The one thing I'm really struggling with is mapping out my dial-plan during my read through of the lab. I would love to hear what others are doing. In my previous attempts, I find it very hard too, because the questions are verbose and I could either spend too much time reading OR not able to encode it into the table form correctly in haste OR simply skipped and waste too much time re-reading it. Here is my plan for my next attempt.I think the key is to have my pre-fabricate table then I will create my table quickly and ONLY adjust it while I read the question. So, I would quickly create this template. The list is there for easy cut & paste. I will only complete Site A in during the lab, then I will just copy to SiteB & SiteC. The rest is just modification of the table. (Note: I find that using "TAB" make it easier to align the columns, it could be 3 or 4 TABs.) In essence, focus on [1] Pre-fabrication [2] Quick to reproduce as a template. The rest is depending on how quick you can decipher verbose question and re-adjust the table. Get the screenshot here is the following format is bad - https://www.dropbox.com/s/o1ftbh4katnhjcu/Quick-Dial-Plan-Template.jpg, The TXT version is here - https://www.dropbox.com/s/m2p2lte5ezk6q7d/Quick-Dial-Plan-Template.TXT I think Matthew Berry youtube is good too - http://www.youtube.com/watch?v=4mP5powuFUM.I just think it can be complement by the cookie-cutting approach to encode the table in Notepad that can be reproduced quickly. === The LIST === ISDN Unknown Subscriber National International Any === SiteA === Calling Called Emer 7D / Unknown / ISDN 7D / Unknown / ISDN Local 7D / Subscriber / ISDN 7D / Subscriber / ISDN LD 10D / National / ISDN 10D / National / ISDN Intl 1+10D / International / ISDN 011! / International / ISDN Another variation of the table format is too cater for TEHO scenario or BACKUP Gateway scenario. === SiteA === Calling Called Emer 7D / Unknown / ISDN 7D / Unknown / ISDN Local 7D / Subscriber / ISDN 7D / Subscriber / ISDN Local(Backup via SiteB) 7D / Subscriber / ISDN 7D / Subscriber / ISDN LD 10D / National / ISDN 10D / National / ISDN Intl 1+10D / International / ISDN 011! / International / ISDN Regards, --Somphol. On Fri, Oct 18, 2013 at 12:22 PM, VanBenschoten, Brian < brian.vanbenscho...@corebts.com> wrote: > I've found the QoS questions are very specific to test a certain area of > knowledge. They are not looking for what we would consider a "best > practice" system wide. I think we could skip setting the DSCP values in > CUCM. > > If you think the question calls for it you can have your class-map match > both AF31 and CS3 for signaling. > > > > *From:* ccie_voice-boun...@onlinestudylist.com [mailto: > ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Bill Hatcher > *Sent:* Thursday, October 17, 2013 3:14 PM > *To:* probert...@gmail.com > *Cc:* ccievoice > *Subject:* Re: [OSL | CCIE_Voice] Base configs and Dial Plan mapping > > > > I agree to setting the service parameters to default first. I was > planning on doing that myself. As to changing the DSCP values, it all > depends on what they ask for in the QoS section of the test is all. > > > > On Thu, Oct 17, 2013 at 2:24 PM, probert...@gmail.com < > probert...@gmail.com> wrote: > > Hi, > > I think my strategy will be to set all Service Parameters to default > before making changes. This way I can avoid and undesirable presets. > > Let me know your thoughts on this. > > Also why are you setting DSCP for Phone Configuration and DSCP for Cisco > CallManager to Device Interface to AF31? Default CS3 should be good, let me > know if I'm wrong on this? > > > > Rob > > > > On Thu, Oct 17, 2013 at 1:39 PM, Bill Hatcher wrote: > > My test is just a couple of weeks away, and I've been reading different > blogs on how to maximize your time. The one thing I'm really struggling > with is mapping out my dial-plan during my read through of the lab. I > would love to hear what others are doing. > > I have also been building base router configs for h323, gatekeeper, mgcp, > srst,sip, etc so that I can practice quickly configuring those on the > routers. > > One of the things I haven't really been keeping track of are some of the > service parameters that I should adjust out of habit. Here are a few that I > can think of off the top of my head that I plan on tweaking at the start of > the exam. Please feel free to add to them. > > Enterprise Parameters > DSCP for Phone Configuration - Set to AF31 > DSCP for Cisco CallMana
Re: [OSL | CCIE_Voice] Base configs and Dial Plan mapping
I've found the QoS questions are very specific to test a certain area of knowledge. They are not looking for what we would consider a "best practice" system wide. I think we could skip setting the DSCP values in CUCM. If you think the question calls for it you can have your class-map match both AF31 and CS3 for signaling. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Bill Hatcher Sent: Thursday, October 17, 2013 3:14 PM To: probert...@gmail.com Cc: ccievoice Subject: Re: [OSL | CCIE_Voice] Base configs and Dial Plan mapping I agree to setting the service parameters to default first. I was planning on doing that myself. As to changing the DSCP values, it all depends on what they ask for in the QoS section of the test is all. On Thu, Oct 17, 2013 at 2:24 PM, probert...@gmail.com<mailto:probert...@gmail.com> mailto:probert...@gmail.com>> wrote: Hi, I think my strategy will be to set all Service Parameters to default before making changes. This way I can avoid and undesirable presets. Let me know your thoughts on this. Also why are you setting DSCP for Phone Configuration and DSCP for Cisco CallManager to Device Interface to AF31? Default CS3 should be good, let me know if I'm wrong on this? Rob On Thu, Oct 17, 2013 at 1:39 PM, Bill Hatcher mailto:wchatc...@gmail.com>> wrote: My test is just a couple of weeks away, and I've been reading different blogs on how to maximize your time. The one thing I'm really struggling with is mapping out my dial-plan during my read through of the lab. I would love to hear what others are doing. I have also been building base router configs for h323, gatekeeper, mgcp, srst,sip, etc so that I can practice quickly configuring those on the routers. One of the things I haven't really been keeping track of are some of the service parameters that I should adjust out of habit. Here are a few that I can think of off the top of my head that I plan on tweaking at the start of the exam. Please feel free to add to them. Enterprise Parameters DSCP for Phone Configuration - Set to AF31 DSCP for Cisco CallManager to Device Interface - Set to AF31 Change the Phone URL's to IP's Organization Top Level Domain Cluster Fully Qualified Domain Name Service Parameters - CallManager T302 Time - Know where it is if you need ot change interdigit timeout Call Classification - Offnet Builtin Bridge Enabled - True Device Name of GK-controlled Trunk That Will Use Port 1720 - Set as needed. Transfer On-hook Enabled - True (Also a great thing to do in production when migrating users from other phone systems) Block offnet to offnet transfers - Know where it's at. Auto Call Pickup Enabled - True Call Back Enabled Flag - True (Verify) Single Button Barge/CBarge Policy - Set to Barge unless otherwise directed. Stop routing on Unallocated Number Flag - False - H323 redundancy Preferred G.711 Millisecond Packet Size - 20 (Verify) Preferred G.729 Millisecond Packet Size - 20 (Verify) G722 Codec Enabled - Disabled (Unless otherwise directed) Intraregion Audio Codec Default - G711/G722 (Verify) Interregion Audio Codec Default - G729 (Verify) Automated Alternate Routing Enabled - True (This one gets me every time on AAR so I turn it on by default now) Enable Mobile Voice Access - Set as required Mobile Voice Access Number - Set as required System Remote Access Blocked Numbers - Set as required Service Parameters -Cisco IP Voice Media Streaming App Supported MOH Codecs - G711 mulaw and G729 Annex A HTH Bill Hatcher ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com<http://www.ipexpert.com> Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com<http://www.PlatinumPlacement.com> Important Notice: This email message and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you are not the named addressee, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. Please note that any views or opinions presented in this email are solely those of the author and do not necessarily represent those of Core BTS. Core BTS specifically disclaims liability for any damage caused by any virus transmitted by this email. ___ 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] Base configs and Dial Plan mapping
Hi, On service parameters, you may also want to check Vik's article http://blog.ipexpert.com/2010/10/13/common-ucm-service-parameters-to-change/ . On the comment section, Trinifox also mentioned "Please add: Intraregion Audio Codec Default to G729 to avoid CSCsl74701 Bug". In my checklist, I also tweak "Conference" section esp. "Drop Ad Hoc Conference". Regards, --Somphol. On Fri, Oct 18, 2013 at 7:14 AM, Bill Hatcher wrote: > I agree to setting the service parameters to default first. I was > planning on doing that myself. As to changing the DSCP values, it all > depends on what they ask for in the QoS section of the test is all. > > > On Thu, Oct 17, 2013 at 2:24 PM, probert...@gmail.com < > probert...@gmail.com> wrote: > >> Hi, >> >> I think my strategy will be to set all Service Parameters to default >> before making changes. This way I can avoid and undesirable presets. >> Let me know your thoughts on this. >> >> Also why are you setting DSCP for Phone Configuration and DSCP for Cisco >> CallManager to Device Interface to AF31? Default CS3 should be good, let me >> know if I'm wrong on this? >> >> Rob >> >> >> On Thu, Oct 17, 2013 at 1:39 PM, Bill Hatcher wrote: >> >>> My test is just a couple of weeks away, and I've been reading different >>> blogs on how to maximize your time. The one thing I'm really struggling >>> with is mapping out my dial-plan during my read through of the lab. I >>> would love to hear what others are doing. >>> >>> I have also been building base router configs for h323, gatekeeper, >>> mgcp, srst,sip, etc so that I can practice quickly configuring those on the >>> routers. >>> >>> One of the things I haven't really been keeping track of are some of the >>> service parameters that I should adjust out of habit. Here are a few that I >>> can think of off the top of my head that I plan on tweaking at the start of >>> the exam. Please feel free to add to them. >>> >>> Enterprise Parameters >>> DSCP for Phone Configuration - Set to AF31 >>> DSCP for Cisco CallManager to Device Interface - Set to AF31 >>> Change the Phone URL's to IP's >>> Organization Top Level Domain >>> Cluster Fully Qualified Domain Name >>> >>> Service Parameters - CallManager >>> T302 Time - Know where it is if you need ot change interdigit timeout >>> Call Classification - Offnet >>> Builtin Bridge Enabled - True >>> Device Name of GK-controlled Trunk That Will Use Port 1720 - Set as >>> needed. >>> Transfer On-hook Enabled - True (Also a great thing to do in production >>> when migrating users from other phone systems) >>> Block offnet to offnet transfers - Know where it's at. >>> Auto Call Pickup Enabled - True >>> Call Back Enabled Flag - True (Verify) >>> Single Button Barge/CBarge Policy - Set to Barge unless otherwise >>> directed. >>> Stop routing on Unallocated Number Flag - False - H323 redundancy >>> Preferred G.711 Millisecond Packet Size - 20 (Verify) >>> Preferred G.729 Millisecond Packet Size - 20 (Verify) >>> G722 Codec Enabled - Disabled (Unless otherwise directed) >>> Intraregion Audio Codec Default - G711/G722 (Verify) >>> Interregion Audio Codec Default - G729 (Verify) >>> Automated Alternate Routing Enabled - True (This one gets me every time >>> on AAR so I turn it on by default now) >>> Enable Mobile Voice Access - Set as required >>> Mobile Voice Access Number - Set as required >>> System Remote Access Blocked Numbers - Set as required >>> >>> Service Parameters -Cisco IP Voice Media Streaming App >>> Supported MOH Codecs - G711 mulaw and G729 Annex A >>> >>> HTH >>> >>> Bill Hatcher >>> >>> ___ >>> 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] Base configs and Dial Plan mapping
I agree to setting the service parameters to default first. I was planning on doing that myself. As to changing the DSCP values, it all depends on what they ask for in the QoS section of the test is all. On Thu, Oct 17, 2013 at 2:24 PM, probert...@gmail.com wrote: > Hi, > > I think my strategy will be to set all Service Parameters to default > before making changes. This way I can avoid and undesirable presets. > Let me know your thoughts on this. > > Also why are you setting DSCP for Phone Configuration and DSCP for Cisco > CallManager to Device Interface to AF31? Default CS3 should be good, let me > know if I'm wrong on this? > > Rob > > > On Thu, Oct 17, 2013 at 1:39 PM, Bill Hatcher wrote: > >> My test is just a couple of weeks away, and I've been reading different >> blogs on how to maximize your time. The one thing I'm really struggling >> with is mapping out my dial-plan during my read through of the lab. I >> would love to hear what others are doing. >> >> I have also been building base router configs for h323, gatekeeper, mgcp, >> srst,sip, etc so that I can practice quickly configuring those on the >> routers. >> >> One of the things I haven't really been keeping track of are some of the >> service parameters that I should adjust out of habit. Here are a few that I >> can think of off the top of my head that I plan on tweaking at the start of >> the exam. Please feel free to add to them. >> >> Enterprise Parameters >> DSCP for Phone Configuration - Set to AF31 >> DSCP for Cisco CallManager to Device Interface - Set to AF31 >> Change the Phone URL's to IP's >> Organization Top Level Domain >> Cluster Fully Qualified Domain Name >> >> Service Parameters - CallManager >> T302 Time - Know where it is if you need ot change interdigit timeout >> Call Classification - Offnet >> Builtin Bridge Enabled - True >> Device Name of GK-controlled Trunk That Will Use Port 1720 - Set as >> needed. >> Transfer On-hook Enabled - True (Also a great thing to do in production >> when migrating users from other phone systems) >> Block offnet to offnet transfers - Know where it's at. >> Auto Call Pickup Enabled - True >> Call Back Enabled Flag - True (Verify) >> Single Button Barge/CBarge Policy - Set to Barge unless otherwise >> directed. >> Stop routing on Unallocated Number Flag - False - H323 redundancy >> Preferred G.711 Millisecond Packet Size - 20 (Verify) >> Preferred G.729 Millisecond Packet Size - 20 (Verify) >> G722 Codec Enabled - Disabled (Unless otherwise directed) >> Intraregion Audio Codec Default - G711/G722 (Verify) >> Interregion Audio Codec Default - G729 (Verify) >> Automated Alternate Routing Enabled - True (This one gets me every time >> on AAR so I turn it on by default now) >> Enable Mobile Voice Access - Set as required >> Mobile Voice Access Number - Set as required >> System Remote Access Blocked Numbers - Set as required >> >> Service Parameters -Cisco IP Voice Media Streaming App >> Supported MOH Codecs - G711 mulaw and G729 Annex A >> >> HTH >> >> Bill Hatcher >> >> ___ >> 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] Base configs and Dial Plan mapping
Hi, I think my strategy will be to set all Service Parameters to default before making changes. This way I can avoid and undesirable presets. Let me know your thoughts on this. Also why are you setting DSCP for Phone Configuration and DSCP for Cisco CallManager to Device Interface to AF31? Default CS3 should be good, let me know if I'm wrong on this? Rob On Thu, Oct 17, 2013 at 1:39 PM, Bill Hatcher wrote: > My test is just a couple of weeks away, and I've been reading different > blogs on how to maximize your time. The one thing I'm really struggling > with is mapping out my dial-plan during my read through of the lab. I > would love to hear what others are doing. > > I have also been building base router configs for h323, gatekeeper, mgcp, > srst,sip, etc so that I can practice quickly configuring those on the > routers. > > One of the things I haven't really been keeping track of are some of the > service parameters that I should adjust out of habit. Here are a few that I > can think of off the top of my head that I plan on tweaking at the start of > the exam. Please feel free to add to them. > > Enterprise Parameters > DSCP for Phone Configuration - Set to AF31 > DSCP for Cisco CallManager to Device Interface - Set to AF31 > Change the Phone URL's to IP's > Organization Top Level Domain > Cluster Fully Qualified Domain Name > > Service Parameters - CallManager > T302 Time - Know where it is if you need ot change interdigit timeout > Call Classification - Offnet > Builtin Bridge Enabled - True > Device Name of GK-controlled Trunk That Will Use Port 1720 - Set as needed. > Transfer On-hook Enabled - True (Also a great thing to do in production > when migrating users from other phone systems) > Block offnet to offnet transfers - Know where it's at. > Auto Call Pickup Enabled - True > Call Back Enabled Flag - True (Verify) > Single Button Barge/CBarge Policy - Set to Barge unless otherwise directed. > Stop routing on Unallocated Number Flag - False - H323 redundancy > Preferred G.711 Millisecond Packet Size - 20 (Verify) > Preferred G.729 Millisecond Packet Size - 20 (Verify) > G722 Codec Enabled - Disabled (Unless otherwise directed) > Intraregion Audio Codec Default - G711/G722 (Verify) > Interregion Audio Codec Default - G729 (Verify) > Automated Alternate Routing Enabled - True (This one gets me every time on > AAR so I turn it on by default now) > Enable Mobile Voice Access - Set as required > Mobile Voice Access Number - Set as required > System Remote Access Blocked Numbers - Set as required > > Service Parameters -Cisco IP Voice Media Streaming App > Supported MOH Codecs - G711 mulaw and G729 Annex A > > HTH > > Bill Hatcher > > ___ > 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] Base configs and Dial Plan mapping
My test is just a couple of weeks away, and I've been reading different blogs on how to maximize your time. The one thing I'm really struggling with is mapping out my dial-plan during my read through of the lab. I would love to hear what others are doing. I have also been building base router configs for h323, gatekeeper, mgcp, srst,sip, etc so that I can practice quickly configuring those on the routers. One of the things I haven't really been keeping track of are some of the service parameters that I should adjust out of habit. Here are a few that I can think of off the top of my head that I plan on tweaking at the start of the exam. Please feel free to add to them. Enterprise Parameters DSCP for Phone Configuration - Set to AF31 DSCP for Cisco CallManager to Device Interface - Set to AF31 Change the Phone URL's to IP's Organization Top Level Domain Cluster Fully Qualified Domain Name Service Parameters - CallManager T302 Time - Know where it is if you need ot change interdigit timeout Call Classification - Offnet Builtin Bridge Enabled - True Device Name of GK-controlled Trunk That Will Use Port 1720 - Set as needed. Transfer On-hook Enabled - True (Also a great thing to do in production when migrating users from other phone systems) Block offnet to offnet transfers - Know where it's at. Auto Call Pickup Enabled - True Call Back Enabled Flag - True (Verify) Single Button Barge/CBarge Policy - Set to Barge unless otherwise directed. Stop routing on Unallocated Number Flag - False - H323 redundancy Preferred G.711 Millisecond Packet Size - 20 (Verify) Preferred G.729 Millisecond Packet Size - 20 (Verify) G722 Codec Enabled - Disabled (Unless otherwise directed) Intraregion Audio Codec Default - G711/G722 (Verify) Interregion Audio Codec Default - G729 (Verify) Automated Alternate Routing Enabled - True (This one gets me every time on AAR so I turn it on by default now) Enable Mobile Voice Access - Set as required Mobile Voice Access Number - Set as required System Remote Access Blocked Numbers - Set as required Service Parameters -Cisco IP Voice Media Streaming App Supported MOH Codecs - G711 mulaw and G729 Annex A HTH Bill Hatcher ___ 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
