Dam I am an idoit... I did not read the question.. no compression needed
-----Original Message----- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Monday, October 04, 2010 9:58 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 56, Issue 47 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. lab 6 QOS question (Leslie Meade) 2. Re: lab 6 QOS question (linuxboss.9) 3. Re: lab 6 QOS question (Leslie Meade) 4. Re: SIP Phones in CME (CCIE Voice GMAIL) ---------------------------------------------------------------------- Message: 1 Date: Mon, 4 Oct 2010 09:50:11 -0700 From: "Leslie Meade" <lme...@signal.ca> To: <ccie_voice@onlinestudylist.com> Subject: [OSL | CCIE_Voice] lab 6 QOS question Message-ID: <65be43a9da05cd44a3a72b458a7c0c5917e...@exch-mg.mgvfs.mcleannet> Content-Type: text/plain; charset="iso-8859-1" I have been looking at the WAN FRF.12 configs and I do not understand something. In all the examples the following has been advised. FRF.12 ( average values) Layer 2 7 bytes Layer 3 2 bytes Media 20 bytes 29 * .008 * 50 = 11.6 Why in this lab it is different FRF.12 Layer 2 8 bytes Layer 3 40 bytes ? ?? Media 20 bytes 68* .008 * 50 = 27.2 I might be missing something but, I do not fully understand where the 40 bytes come into play. -------------- next part -------------- An HTML attachment was scrubbed... URL: </archives/ccie_voice/attachments/20101004/0ad84d56/attachment-0001.html> ------------------------------ Message: 2 Date: Mon, 4 Oct 2010 09:52:43 -0700 From: "linuxboss.9" <linuxbos...@gmail.com> To: Leslie Meade <lme...@signal.ca> Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] lab 6 QOS question Message-ID: <aanlktinrddbwodk-bgp3g-g5yxsrprj4dqpesojk1...@mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" RTP + UDP headers = 40 bytes On Mon, Oct 4, 2010 at 9:50 AM, Leslie Meade <lme...@signal.ca> wrote: > I have been looking at the WAN FRF.12 configs and I do not understand > something. > > > > In all the examples the following has been advised. > > > > FRF.12 ( average values) > > Layer 2 7 bytes > > Layer 3 2 bytes > > Media 20 bytes > > > > 29 * .008 * 50 = 11.6 > > > > Why in this lab it is different > > > > FRF.12 > > Layer 2 8 bytes > > Layer 3 40 bytes ? ?? > > Media 20 bytes > > 68* .008 * 50 = 27.2 > > > > I might be missing something but, I do not fully understand where the > 40 bytes come into play. > > > > _______________________________________________ > For more information regarding industry leading CCIE Lab training, > please visit www.ipexpert.com > > -------------- next part -------------- An HTML attachment was scrubbed... URL: </archives/ccie_voice/attachments/20101004/c8772396/attachment-0001.html> ------------------------------ Message: 3 Date: Mon, 4 Oct 2010 09:57:26 -0700 From: "Leslie Meade" <lme...@signal.ca> To: "linuxboss.9" <linuxbos...@gmail.com> Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] lab 6 QOS question Message-ID: <65be43a9da05cd44a3a72b458a7c0c5917e...@exch-mg.mgvfs.mcleannet> Content-Type: text/plain; charset="iso-8859-1" Ok given that that's the reason, why is it not included in previous examples ? For example, I have seen this example given lots of times.... Ip/udp/rtp 2 bytes /packet From: linuxboss.9 [mailto:linuxbos...@gmail.com] Sent: Monday, October 04, 2010 9:53 AM To: Leslie Meade Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] lab 6 QOS question RTP + UDP headers = 40 bytes On Mon, Oct 4, 2010 at 9:50 AM, Leslie Meade <lme...@signal.ca> wrote: I have been looking at the WAN FRF.12 configs and I do not understand something. In all the examples the following has been advised. FRF.12 ( average values) Layer 2 7 bytes Layer 3 2 bytes Media 20 bytes 29 * .008 * 50 = 11.6 Why in this lab it is different FRF.12 Layer 2 8 bytes Layer 3 40 bytes ? ?? Media 20 bytes 68* .008 * 50 = 27.2 I might be missing something but, I do not fully understand where the 40 bytes come into play. _______________________________________________ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -------------- next part -------------- An HTML attachment was scrubbed... URL: </archives/ccie_voice/attachments/20101004/59a06672/attachment-0001.html> ------------------------------ Message: 4 Date: Mon, 4 Oct 2010 09:58:05 -0700 From: "CCIE Voice GMAIL" <givemeccievoice2...@gmail.com> To: "'OSL Group'" <ccie_voice@onlinestudylist.com> Subject: Re: [OSL | CCIE_Voice] SIP Phones in CME Message-ID: <005e01cb63e5$514e2ed0$f3ea8c...@com> Content-Type: text/plain; charset="us-ascii" Thanks Vik. I have reset the scenario, however if I run into the problem again I will try the factory reset. Jeff From: Vik Malhi [mailto:vma...@ipexpert.com] Sent: Monday, October 04, 2010 9:17 AM To: CCIE Voice GMAIL; OSL Group Subject: Re: [OSL | CCIE_Voice] SIP Phones in CME I would press the red button and restore fac defaults - after a power cycle hold down # and then once you see the blinking lights 123456789*0# -- Vik Malhi - CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat <http://www.ipexpert.com/chat> IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (R&S, Voice, Wireless, Security & Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities <http://www.ipexpert.com/communities> and our public website at www.ipexpert.com <http://www.ipexpert.com/> _____ From: CCIE Voice GMAIL <givemeccievoice2...@gmail.com> Date: Mon, 4 Oct 2010 09:05:38 -0700 To: OSL Group <ccie_voice@onlinestudylist.com> Subject: Re: [OSL | CCIE_Voice] SIP Phones in CME Hi Vik, Is that done by pressing "**#**" and then selecting the "Erase" softkey? I've done that a couple times, and actually I did that right before this problem started. I figured it may have had something to do with the problem, however I didn't want to make assumptions. jeff From: Vik Malhi [mailto:vma...@ipexpert.com] Sent: Monday, October 04, 2010 8:47 AM To: CCIE Voice GMAIL; OSL Group Subject: Re: [OSL | CCIE_Voice] SIP Phones in CME I've seen this problem whereby it does not progress from grabbling the cnf file. The only thing I have ever been able to do to stop this endless cycle is to erase the cnf file from the phone. -- Vik Malhi - CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat <http://www.ipexpert.com/chat> IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (R&S, Voice, Wireless, Security & Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities <http://www.ipexpert.com/communities> and our public website at www.ipexpert.com <http://www.ipexpert.com/> _____ From: CCIE Voice GMAIL <givemeccievoice2...@gmail.com> Date: Sun, 3 Oct 2010 18:41:28 -0700 To: OSL Group <ccie_voice@onlinestudylist.com> Subject: Re: [OSL | CCIE_Voice] SIP Phones in CME Hi Randall, Thanks for the input. I have since re-loaded the scenario and started anew. However, I did verify on the phones that they correct default gateway was being issued. Also, the phones are connected to a ESW module local to the router that was playing the role of CME. There was a voice vlan that was up and two FastEthernet ports that were also up, otherwise the phones wouldn't be on. I'm going to try to do this scenario again at some point this week, so if I run into the same problem again I will be reaching out to you guys for help J Jeff From: Randall Saborio [mailto:ill2...@gmail.com] Sent: Sunday, October 03, 2010 2:02 PM To: CCIE Voice GMAIL Cc: osl osl Subject: Re: [OSL | CCIE_Voice] SIP Phones in CME Jeff, The only other thing I could think of is that one of your interfaces must have been down, or the IP Phones didn't have a proper default gateway. Cause your telephony config seems allright, but then if you don't get any sip messages, then it should be related to network configuration. Not sure if you managed something already. Cheers. On Fri, Oct 1, 2010 at 8:54 PM, CCIE Voice GMAIL <givemeccievoice2...@gmail.com> wrote: I see now after looking again that the IP address with Name is for the NTP server that I have configured under voice register global. From: CCIE Voice GMAIL [mailto:givemeccievoice2...@gmail.com] Sent: Friday, October 01, 2010 7:50 PM To: 'Randall Saborio' Cc: 'osl osl' Subject: RE: [OSL | CCIE_Voice] SIP Phones in CME Hi Randall, I actually did the delete command previously. Then after I did a "no create profile" CME stated that it couldn't delete the file b/c it wasn't there. Then I issued a "create profile". Here is the output from the more command. I've bolded what I felt was the important items. The only thing that stands out is the whole "name" tag, as I'm not sure why it has a different router's IP address. Other than that, as far as I can tell, this is all the right information. <device> <deviceProtocol>SIP</deviceProtocol> <devicePool> <dateTimeSetting> <dateTemplate>Y/M/DA</dateTemplate> <timeZone>Pacific Standard/Daylight Time</timeZone> <ntps> <ntp priority="0"> <name>10.5.200.1</name> <ntpMode>directedbroadcast</ntpMode> </ntp> </ntps> </dateTimeSetting> <callManagerGroup> <members> <member priority="0"> <callManager> <ports> <sipPort>5060</sipPort> </ports> <processNodeName>10.5.202.1</processNodeName> </callManager> </member> </members> </callManagerGroup> </devicePool> <sipProfile> <sipProxies> <registerWithProxy>true</registerWithProxy> </sipProxies> <sipCallFeatures> <cnfJoinEnabled>true</cnfJoinEnabled> <localCfwdEnable>true</localCfwdEnable> <callForwardURI>service-uri-cfwdall</callForwardURI> <callPickupURI>service-uri-pickup</callPickupURI> <callPickupGroupURI>service-uri-gpickup</callPickupGroupURI> <callHoldRingback>2</callHoldRingback> <semiAttendedTransfer>true</semiAttendedTransfer> <anonymousCallBlock>2</anonymousCallBlock> <callerIdBlocking>2</callerIdBlocking> <dndControl>2</dndControl> <remoteCcEnable>true</remoteCcEnable> </sipCallFeatures> <sipStack> <remotePartyID>true</remotePartyID> </sipStack> <sipLines> <line button="1"> <featureID>9</featureID> <featureLabel>4001</featureLabel> <proxy>USECALLMANAGER</proxy> <port>5060</port> <name>4001</name> <displayName>Site C Phone 1</displayName> <autoAnswer> <autoAnswerEnabled>2</autoAnswerEnabled> </autoAnswer> <callWaiting>1</callWaiting> <authName>scuser1</authName> <authPassword>cisco</authPassword> <sharedLine>false</sharedLine> <messagesNumber>4500</messagesNumber> <ringSettingActive>5</ringSettingActive> <forwardCallInfoDisplay> <callerName>true</callerName> <callerNumber>true</callerNumber> <redirectedNumber>true</redirectedNumber> <dialedNumber>true</dialedNumber> </forwardCallInfoDisplay> </line> <line button="2"> <featureID>21</featureID> <featureLabel>SCPH2 4002</featureLabel> <speedDialNumber>4002</speedDialNumber> </line> </sipLines> <enableVad>true</enableVad> <preferredCodec>g711alaw</preferredCodec> <softKeyFile>softkeyDefault_kpml.xml</softKeyFile> <dialTemplate></dialTemplate> <kpml>1</kpml> <phoneLabel>+442321314001</phoneLabel> <stutterMsgWaiting>2</stutterMsgWaiting> <disableLocalSpeedDialConfig>true</disableLocalSpeedDialConfig> <dscpForAudio>184</dscpForAudio> <dscpVideo>136</dscpVideo> </sipProfile> <commonProfile> <phonePassword>cisco</phonePassword> <callLogBlfEnabled>3</callLogBlfEnabled> </commonProfile> <loadInformation>SIP42.9-0-3S</loadInformation> <versionStamp>0004823422355311</versionStamp> <userLocale> <name>English_United_States</name> <langCode>en</langCode> </userLocale> <networkLocale>United_States</networkLocale> <networkLocaleInfo> <name>United_States</name> </networkLocaleInfo> <authenticationURL></authenticationURL> <directoryURL>http://10.5.202.1/localdirectory <http://10.5.202.1/localdirectory%3c/directoryURL%3e> </directoryURL> <http://10.5.202.1/localdirectory%3c/directoryURL%3e> <servicesURL></servicesURL> <dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices> <dscpForCm2Dvce>96</dscpForCm2Dvce> <transportLayerProtocol>2</transportLayerProtocol> </device> From: Randall Saborio [mailto:ill2...@gmail.com] Sent: Friday, October 01, 2010 7:15 PM To: CCIE Voice GMAIL Cc: osl osl Subject: Re: [OSL | CCIE_Voice] SIP Phones in CME Jeff, Unprovisiones just stands for unregistered, so it is a very broad error. Since you say that you don't get any messages when debugging ccsip, I advise the following: - Verify the cnf files on flash actually have the ip for the CME. Just do a "more flash:<filename>" - If it shows the info is not the correct one, I assume the files are there from a previous CME SCCP configuration. Then do a delete flash:<filename>, and then do a create profile again. - I believe once the phone cannot find the SEP<mac> file, then it may attempt to download some other config file that has the updated information for connecting to CME as SIP. HTH. On Fri, Oct 1, 2010 at 8:09 PM, CCIE Voice GMAIL <givemeccievoice2...@gmail.com> wrote: Well I did the below steps and I'm still getting an "Unprovisioned" message. This is frustrating as it was working before. Any other ideas? I'll keep everyone posted as I see in the archives I'm not the only one who has experienced this problem. Jeff From: CCIE Voice GMAIL [mailto:givemeccievoice2...@gmail.com] Sent: Friday, October 01, 2010 6:21 PM To: 'Daniel Berlinski' Cc: 'osl osl' Subject: RE: [OSL | CCIE_Voice] SIP Phones in CME Hi Dan, I am not seeing any output from debug ccsip messages. I am not using proctor labs, so I must be using a different version of CME (I believe 7), because presence is working for me (I guess I should say was ..). I read in the archives that someone used CUCM to do the following and it worked: - Convert to SCCP - Migrate to SIP - No create profile - Create profile - Register with CME I'm going to give that shot and see if this works. Jeff From: Daniel Berlinski [mailto:dberlin...@gmail.com] Sent: Friday, October 01, 2010 6:01 PM To: CCIE Voice GMAIL Subject: Re: [OSL | CCIE_Voice] SIP Phones in CME Hi Please try the following: Ensure your ntp server is synced no create profile to delete all config file and then create profile to recreate them If still no dice then please check with debug ccsip messages for authorization error. If you see authentication/authorization probs please use the primary line number as the username i.e. username 4001 password cisco. Other things to note: CME presence is not supported in SIP phones in this version of CME so you do not need those commands Although voice-class codec is allowed to be entered the phone will only support one codec type which is g711ula, so voice-class codec is not helping you either. Try to simplify a bit your configs until you get your phones registered. Cheers On Sat, Oct 2, 2010 at 1:51 PM, CCIE Voice GMAIL <givemeccievoice2...@gmail.com> wrote: I added that command: voice register global mode cme source-address 10.5.202.1 port 5060 max-dn 20 max-pool 2 load 7945 SIP45.9-0-3S load 7942 SIP42.9-0-3S authenticate register date-format Y/M/D voicemail 4500 url directory http://10.5.202.1/localdirectory tftp-path flash: create profile sync 0001302544054016 ntp-server 10.5.200.1 mode directedbroadcast voice register dn 1 number 4001 call-forward b2bua busy 4500 call-forward b2bua noan 4500 timeout 10 allow watch name Site C Phone 1 label 4001 voice register dn 2 number 4002 call-forward b2bua busy 4500 call-forward b2bua noan 4500 timeout 10 allow watch name Site C Phone 2 label 4002 voice register pool 1 id mac 0024.9733.6C28 type 7942 number 1 dn 1 presence call-list dtmf-relay rtp-nte sip-notify voice-class codec 1 username scuser1 password cisco description +442321314001 blf-speed-dial 1 4002 label "SCPH2 4002" device privacy off voice register pool 2 id mac 0024.14B2.F542 type 7945 number 1 dn 2 presence call-list dtmf-relay rtp-nte sip-notify voice-class codec 1 username scuser2 password cisco description +442321314002 blf-speed-dial 1 4001 label "SCPH1 4001" device privacy off And I did a debug tftp events to see what was happening. It appears like the phones are accessing their configuration files correctly: .Oct 2 02:05:19.602: TFTP: Looking for SEP002497336C28.cnf.xml .Oct 2 02:05:19.606: TFTP: Opened flash:/SEP002497336C28.cnf.xml, fd 7, size 3046 for process 363 .Oct 2 02:05:19.610: TFTP: Finished flash:/SEP002497336C28.cnf.xml, time 00:00:00 for process 363 R3(config-if-range)# .Oct 2 02:05:35.362: TFTP: Looking for CTLSEP002414B2F542.tlv .Oct 2 02:05:35.462: TFTP: Looking for SEP002414B2F542.cnf.xml .Oct 2 02:05:35.466: TFTP: Opened flash:/SEP002414B2F542.cnf.xml, fd 7, size 3046 for process 363 .Oct 2 02:05:35.474: TFTP: Finished flash:/SEP002414B2F542.cnf.xml, time 00:00:00 for process 363 I am still seeing the phone say "Unprovisioned" and the status messages that say "Error Verifying Config Info". I have done a "create profile" after every change I've made, so I'm not sure what's going on here L Any other ideas? Thank you Dan for all of your input. Jeff From: Daniel Berlinski [mailto:dberlin...@gmail.com] Sent: Friday, October 01, 2010 5:30 PM To: CCIE Voice GMAIL Cc: osl osl Subject: Re: [OSL | CCIE_Voice] SIP Phones in CME I beleive you are missing tftp path flash: under voice register global. Can you try, create profile and let us know? On Sat, Oct 2, 2010 at 1:11 PM, CCIE Voice GMAIL <givemeccievoice2...@gmail.com> wrote: Hi again, I actually reloaded my router with clean configuration and then re-configured CME, however I am still seeing the same problem. I erased the configurations on the phones before this all happened, so I assume this is maybe part of the problem. I don't know why it would be though, as the phones are getting IP addresses from DHCP and communicating with CME. This is my relevant configs: < ----- DHCP FOR PHONES ----- > ip dhcp excluded-address 10.5.202.1 ip dhcp pool SC_PHONES network 10.5.202.0 255.255.255.0 option 150 ip 10.5.202.1 default-router 10.5.202.1 < ----- VOICE SERVICE ----- > voice service voip allow-connections sip to sip fax protocol cisco sip bind control source-interface Vlan250 bind media source-interface Vlan250 registrar server expires max 1200 min 500 < ----- VOICE CODEC ----- > voice class codec 1 codec preference 1 g711alaw codec preference 2 g711ulaw codec preference 3 g729r8 < ----- SIP CME CONFIG ----- > voice register global mode cme source-address 10.5.202.1 port 5060 max-dn 20 max-pool 2 load 7945 SIP45.9-0-3S load 7942 SIP42.9-0-3S authenticate register date-format Y/M/D voicemail 4500 url directory http://10.5.202.1/localdirectory create profile sync 0001302544054013 ntp-server 10.5.200.1 mode directedbroadcast < ----- PHONE 1 LINE 1 ----- > voice register dn 1 number 4001 call-forward b2bua busy 4500 call-forward b2bua noan 4500 timeout 10 allow watch name Site C Phone 1 label 4001 < ----- PHONE 2 LINE 1 ----- > voice register dn 2 number 4002 call-forward b2bua busy 4500 call-forward b2bua noan 4500 timeout 10 allow watch name Site C Phone 2 label 4002 < ----- PHONE 1 (7942) ----- > voice register pool 1 id mac 0024.9733.6C28 type 7942 number 1 dn 1 presence call-list dtmf-relay rtp-nte sip-notify voice-class codec 1 username scuser1 password cisco description +442321314001 blf-speed-dial 1 4002 label "SCPH2 4002" device privacy off < ----- PHONE 2 (7945) ----- > voice register pool 2 id mac 0024.14B2.F542 type 7945 number 1 dn 2 presence call-list dtmf-relay rtp-nte sip-notify voice-class codec 1 username scuser2 password cisco description +442321314002 blf-speed-dial 1 4001 label "SCPH1 4001" device privacy off < ----- TFTP FILES ----- > tftp-server flash:SIP/apps42.9-0-3TH1-22.sbn alias apps42.9-0-3TH1-22.sbn tftp-server flash:apps42.9-0-3TH1-22.sbn alias cnu42.9-0-3TH1-22.sbn tftp-server flash:SIP/cvm42sip.9-0-3TH1-22.sbn alias cvm42sip.9-0-3TH1-22.sbn tftp-server flash:SIP/dsp42.9-0-3TH1-22.sbn alia _____ _______________________________________________ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -------------- next part -------------- An HTML attachment was scrubbed... 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