Re: [OSL | CCIE_Voice] Setting number plan indicator on the dial peer without a translation rule
Hi all, This command here is used to match the number type, not to set it like in voice translation rules Thanks and best regards, EHAB SALEM Cisco Instructor | Sigma IT - Egypt From: Angel Perez [mailto:gorr...@hotmail.com] Sent: Wednesday, June 09, 2010 8:32 PM To: ciscovoiceg...@gmail.com; osl osl Subject: Re: [OSL | CCIE_Voice] Setting number plan indicator on the dial peer without a translation rule Hi: I tested it some time ago an it didn't works... so I needed to use voice translation... I think that other people had problems with this also Give it a try a let us know hth _ Date: Tue, 8 Jun 2010 20:36:49 -0500 From: ciscovoiceg...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Setting number plan indicator on the dial peer without a translation rule Reading the Implementing Cisco Voice Gateways and Gatekeepers student guide, page 290. They cite another way to set numbering plan on a dial peer. Here is their example: dial-peer voice 100 pots numbering-type national destination-pattern 91408... prefix 1408 port 1/0:23 Has anyone tried this before? This might be a way to avoid (if needed) setting the type via a translation-rule/profile. Thoughts? -- Matthew Berry A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written Vitals: GVoice: +1.612.424.5044 Gmail: ciscovoiceg...@gmail.com Skype: ciscovoiceguru Twitter: ciscovoiceguru Cert Stats: Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 _ Hotmail: Powerful Free email with security by Microsoft. Get it now. https://signup.live.com/signup.aspx?id=60969 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Passed CCIE Voice ( # 26129 )
Congratulations Moataz, from success to success J Thanks and best regards, EHAB SALEM Cisco Instructor/Consultant (CCIE#26088 Voice) | Sigma IT - Egypt From: Le Minh Khoi [mailto:leminhk...@gmail.com] Sent: Sunday, June 06, 2010 12:36 PM To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Passed CCIE Voice ( # 26129 ) Congrats!!! On 6/5/2010 10:02 PM, Moataz Mamdouh wrote: ALL I passed the exam from the first attempt , thank you all for you help , really i learned a lot from you. Regards Moataz Mamdouh Tolba SEEGYPT Website:www.seegypt.com Technical Support Engineer CCIE # 26129 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] I Passed CCIE#26088!!!
Features and Services guide and SRND for each application (CUCM, CUCME, CUE, UCCX, CUPS and CUC) Then start with IP Expert volume 1 (Technology labs) and volume 2 (Full Scenarios) Thanks and best regards, EHAB SALEM Cisco Instructor | Sigma IT - Egypt From: cisco voip [mailto:voip.ccieci...@gmail.com] Sent: Thursday, May 27, 2010 6:33 AM To: Ehab Salem Subject: Re: [OSL | CCIE_Voice] I Passed CCIE#26088!!! Thanks Ehab, What all guides did you follow for your preperation On Thu, May 27, 2010 at 4:42 AM, Ehab Salem esa...@sigma-it.net wrote: Hi, The most important part that needs a very clear strategy.is the call routing, it shouldn't take more than 1 hour (reading, planning and configuration) then max. 30 mins testing and troubleshooting. I used to build a table before starting the configuration and to have a clear and fixed naming conventions for Voice Translation rules, dial-peers and Rout Lists.etc Also I used to have a fixed strategy in building call routing on CUCM like: when he needs redundancy for a call, I'm configuring a route list containing both gateways and do all the digit manipulation on this Route List, not in Route Pattern. This is what I meant by strategy and plan.also don't forget, sleep well before the exam J Best of luck. Thanks and best regards, EHAB SALEM Cisco Instructor | Sigma IT - Egypt From: cisco voip [mailto:voip.ccieci...@gmail.com] Sent: Wednesday, May 26, 2010 5:52 AM To: Ehab Salem Subject: Re: [OSL | CCIE_Voice] I Passed CCIE#26088!!! Congrats Ehab, This is the first time i saw someone clearing this exam in first attempt. I have my exam on 12th August and even i want to nail it in first attempt. If you can guide me it would be of great help, what was your study plan and what guides did you follow for practice. Thanks On Wed, May 26, 2010 at 3:48 AM, Ehab Salem esa...@sigma-it.net wrote: Dear Group, I Passed from the first shot Really thanks a lot for all your help.I really learned a lot from this kind study list J All what I want to say about my experience: the exam is easier than what we have in Volume 2.so it's all about Time Management, Strategy and Plan. I finished the lab in almost 6 hours. And spent the rest of time revising my configuration. I spent the week before the exam practicing on time management and putting a strategy and plan for each part in the exam that may come..and before the exam you should sleep well to start the exam with your full performance and energy. Anyway, it's over now for me.and wish u all the best J Thanks and best regards, EHAB SALEM Cisco Instructor | Sigma IT - Egypt ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] I Passed CCIE#26088!!!
Dear Group, I Passed from the first shot Really thanks a lot for all your help.I really learned a lot from this kind study list J All what I want to say about my experience: the exam is easier than what we have in Volume 2.so it's all about Time Management, Strategy and Plan. I finished the lab in almost 6 hours. And spent the rest of time revising my configuration. I spent the week before the exam practicing on time management and putting a strategy and plan for each part in the exam that may come..and before the exam you should sleep well to start the exam with your full performance and energy. Anyway, it's over now for me.and wish u all the best J Thanks and best regards, EHAB SALEM Cisco Instructor | Sigma IT - Egypt ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Not getting PLUS on my Phones
You can prefix plus to the incoming calling number from Gateway page in CUCM. Ehab M. Salem From: Ashar Siddiqui [mailto:siddas...@gmail.com] Sent: Sunday, May 23, 2010 4:29 PM To: Wael Agina Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Not getting PLUS on my Phones I am glad you mentioned those rules...I have tried all those rules before exactly...no joy...this is why I wrote in my last email that I tried all my translation rule skills.. :) voice translation-rule 99 rule 1 /^34\(.*\)/ /+34\1/ type any unknown plan any unknown rule 2 // /+/ rule 3 // /+/ type any unknown plan any unknown rule 4 /^34/ /+\0/ Even after all this... R2# May 23 12:27:59.440: ISDN Se0/0/0:23 Q931: RX - SETUP pd = 8 callref = 0x00AB Bearer Capability i = 0x9090A2 Standard = CCITT Transfer Capability = 3.1kHz Audio Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Progress Ind i = 0x8583 - Origination address is non-ISDN Display i = 'SCPH1' Calling Party Number i = 0x0080, '3432143001' Plan:Unknown, Type:Unknown Called Party Number i = 0x80, '6178631001' Plan:Unknown, Type:Unknown May 23 R2# 12:27:59.440: ISDN Se0/0/0:23 Q931: Received SETUP callref = 0x80AB callID = 0x0019 switch = primary-ni interface = User May 23 12:27:59.460: ISDN Se0/0/0:23 Q931: TX - CALL_PROC pd = 8 callref = 0x80AB Channel ID i = 0xA98381 Exclusive, Channel 1 May 23 12:27:59.588: ISDN Se0/0/0:23 Q931: TX - ALERTING pd = 8 callref = 0x80AB R2# May 23 12:28:01.972: ISDN Se0/0/0:23 Q931: RX - DISCONNECT pd = 8 callref = 0x00AB Cause i = 0x8290 - Normal call clearing May 23 12:28:01.976: ISDN Se0/0/0:23 Q931: TX - RELEASE pd = 8 callref = 0x80AB May 23 12:28:01.988: ISDN Se0/0/0:23 Q931: RX - RELEASE_COMP pd = 8 callref = 0x00AB Thanks, Ash Wael Agina wrote: Try This and keep us updated 1- voice translation-rule 99 rule 1 // /+\0/ 2- If above working then make it specific for 34* numbers voice translation-rule 99 rule 1 /^34/ /+\0/ 3- Last resort try num-exp === this will affect both direction calls and any calling passing the router !!! num-exp 3432143... +3432143... ! Thanks and Best Regards, Wael Agina ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Frame relay traffic shaping
If you are configuring FRF.12 you have to write frame-relay traffic shaping under the physical interface to work properly. Thanks and best regards, EHAB SALEM Cisco Instructor | Sigma IT - Egypt From: Omotayo [mailto:adefilabi...@gmail.com] Sent: Monday, May 10, 2010 6:38 AM To: bkvalent...@gmail.com Cc: OSL Group Subject: Re: [OSL | CCIE_Voice] Frame relay traffic shaping thanks On Sun, May 9, 2010 at 7:33 PM, bkvalent...@gmail.com bkvalent...@gmail.com wrote: I wouldn't waste time configuring anything that won't earn you points. The test isn't about how closely you follow best practices. Do what is asked. If you aren't sure what is being asked, check with the proctor. Brian - Reply message - From: Omotayo adefilabi...@gmail.com Date: Sun, May 9, 2010 9:09 pm Subject: [OSL | CCIE_Voice] Frame relay traffic shaping To: OSL Group ccie_voice@onlinestudylist.com Hello all, If questions does not explicitly say we should configure shaping. Do we have to on the physical interface? thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] (no subject)
Go to DHCP settings and make it Manual First. From: asif raza [mailto:asifraz...@hotmail.com] Sent: Friday, April 02, 2010 9:18 AM To: CCIE-Voice Subject: [OSL | CCIE_Voice] (no subject) Dear Friends I want to Edit IP Address of my Cisco IP Phone Model 7911, when I go to Setting and and press **# to unlock phone setting, It do unlocl phone setting but when I go to IP Address section, The Edit Soft key appears but it is disabled. Any Idea how edit IP address in this scnario?? plz... Best Regards Asif _ The New Busy is not the old busy. Search, chat and e-mail from your inbox. Get started. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON: WL:en-US:WM_HMP:042010_3 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol1 lab 9.4
I think you are right Sergio…it should be a mistake. Thanks. From: Sergio Polizer [mailto:spoli...@hotmail.com] Sent: Monday, January 18, 2010 7:10 PM To: esa...@sigma-it.net; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Vol1 lab 9.4 I saw that too . It looks like he made a call from Assistant Primary Line (5002) to Manager and It was automatic diverted to the Proxy Line (1560). There is no logic in the real life, but it’s a way to test if we have just two phones (Manager and Assistant) near you. _ From: esa...@sigma-it.net To: ccie_voice@onlinestudylist.com Date: Sun, 17 Jan 2010 17:15:43 +0200 Subject: [OSL | CCIE_Voice] Vol1 lab 9.4 Hi all, In volume 1 IPMA lab 9.4 he needs to have this console view in case any caller calls the Manager DN and he was diverting all calls to the Assistant: I have the calls not forwarded to 5002, but they are forwarded directly to the proxy line “1560”…so how to have the above screen? Thanks for your help. Ehab M. Salem _ Quer 25 GB de armazenamento gratuito na web? Conheça o Skydrive clicando aqui. http://www.eutenhomaisnowindowslive.com.br/?utm_source=MSN_Hotmailutm_medi um=Taglineutm_campaign=InfuseSocial ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Vol1 Lab 11 Unity Connection in SRST
Hi All, He asked to forward all unanswered calls to BR1Phone1 while in SRST to his Voice mail.the issue I have is: the call is successfully transferred to the voice mail, but the DTMF is not sent to the unity connection. But if I called the voice mail directly from BR1Phone1, the DTMF is sent normally!!! Any idea? Thanks. Ehab M. Salem ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Vol1 lab 9.4
Hi all, In volume 1 IPMA lab 9.4 he needs to have this console view in case any caller calls the Manager DN and he was diverting all calls to the Assistant: I have the calls not forwarded to 5002, but they are forwarded directly to the proxy line 1560.so how to have the above screen? Thanks for your help. Ehab M. Salem image004.jpg___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] subscriber error
You can unplug the Publisher from the network and ask the subscriber to ignore the publisher if he didn't find it. But I don't know if they will synchronize after that or not..you can try Ehab Salem From: anupam TYAGI [mailto:anuf...@gmail.com] Sent: Saturday, December 05, 2009 3:44 PM To: Ehab Salem Subject: Re: [OSL | CCIE_Voice] subscriber error any workaround , On Sat, Dec 5, 2009 at 7:12 PM, Ehab Salem esa...@sigma-it.net wrote: Yes, this is a common problem when installing CUCM7.0 on a VMWare.try CUCM7.1 it will work. Ehab Salem From: anupam TYAGI [mailto:anuf...@gmail.com] Sent: Saturday, December 05, 2009 3:32 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] subscriber error Hi , I have added the server in the ccm piblisher and entered the name and pasword of the first node correctly . I am getting the attached error . Can some one help on this Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 1 Lab 6 - AAR not being activated (UNCLASSIFIED)
Route pattern should be 9.16178631XXX as you made the AAR Group prefix 91 Ehab From: Girard, Jeffrey COL MIL USA [mailto:jeffrey.gir...@us.army.mil] Sent: Wednesday, November 25, 2009 6:37 AM To: o...@ipexpert.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 6 - AAR not being activated (UNCLASSIFIED) 1 - BR1 Phone 2 has external phone number mask set to 6178631XXX with no AAR destination mask set 2 - BR1 Phone 2 has AAR group AAR_HQ_BR1 configured on both the device and the line 3 - HQ Phone 2 has AAR group AAR_HQ_BR1 configured on both the device and the line 4 - The dial prefix for the only (single) AAR group AAR_HQ_BR1 is set to 91 5 - Route Pattern 9.617XXX in the PT_AAR partition points to the roite list RL_HQ. Inside that RL is the RG_HQ which contains the HQ H323 GW (originally started life as a SIP GW) 6 - Reset both phones, rebooted routers, rebooted CUCM servers Jeff _ From: Otto Sanchez o...@ipexpert.com To: Girard, Jeffrey COL MIL USA Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Tue Nov 24 19:42:01 2009 Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 6 - AAR not being activated (UNCLASSIFIED) Make sure of the following (hqp2 to br1p2 call): 1.- br1p2 has an external phone number mask or aar destination mask of 617863 2.- br1p2 Line (1002) has the br1 aar group configured 3.- hqp2 Device has the hq aar group configured 4.- Dial prefix (in AAR Group configuration) from hq aar to br1 aar group equals 91 5.- 91617863 route pattern in PT_AAR points to hq gw 6.- Reset both devices HTH, On Tue, Nov 24, 2009 at 7:01 PM, Girard, Jeffrey COL MIL USA jeffrey.gir...@us.army.mil wrote: Classification: UNCLASSIFIED Caveats: FOUO I have searched the archives and several folks have had this problem, but there is not posting of the solution. What I have done: CUCM pub and sub have clusterwide AAR service set to true Created an AAR group that prepends 91 Created the new CSS/PT/translation patterns and route patterns as per the PG - Route pattern 9.1617XXX inside PT_AAR which is seen by CSS_AAR. This is for HQ Phone 2 (5002) to call BR1 Phone 2 (1002). - Xlation pattern 91212XXX inside PT_AAR which is seen by CSS_AAR. In this xlation pattern, I am setting the use of the external phone number mask to meet the ANI requirements of the question. This xlation pattern has a new CSS of CSS_AAR_post_translate_ANI which sees a partition of PT_AAR_post_translate_ANI. Inside of this partition is the route pattern of 9.1212XXX From HQ Phone 2 (5002) - I am able to call BR 1 Phone 2 (1002) using a Long Distance pattern of 916178631002. I am also able to call using 4 digit dialing of 1002 If I force AAR by reducing the RSVP down to 39, and dial 1002 from the HQ phone, I get the expected Not enough bandwidth message, but not the expected Rerouting message. Debug isdn q931 on the HQ GW shows no activity trying to go out the PSTN link. I tried putting the HQ SIP trunk and the BR1 MGCP GW in the same AAR as the phones, but no change. I have done no MGCP/MGCP For the folks who had this same problem, how did you solve it? Jeff --- Jeffrey T. Girard (Jeff) COL, 53 Future Forces Integration Directorate (FFID), Deputy - Networks office: (915)568-1240 DSN 978 Mobile: (915)727-4222 reply to: jeffrey.gir...@us.army.mil Classification: UNCLASSIFIED Caveats: FOUO ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com blockedhttp://www.ipexpert.com/ -- Regards, Otto Sanchez CCIE #25592 (Voice), CCVP,CCSP,CCNP,CCDA,MCSE. Sr. Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com blockedhttp://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 1 Lab 5.8 Testing failover to MGCP
Dear Guys,
It worked with me by just shutting down the voice-port and disabled the
Stop Routing on javascript:getHelp('StopRoutingOnUnallocatedNumberFlag')
Unallocated Number Flag
Ehab
From: Girard, Jeffrey COL MIL USA [mailto:jeffrey.gir...@us.army.mil]
Sent: Saturday, October 31, 2009 4:26 AM
To: drodrig...@fidelus.com; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 5.8 Testing failover to MGCP
I think that I have shown that the up/down status of the voice port does not
effect the SIP trunk status from the CUM point of view. CUCM still sees it
as up and still tries to route across it
Believe that I have found an error in the PG
For those that have not goteen to 5.8 yet - I added a solution that worked
foe me
Jeff
- Original Message -
From: Daniel Rodriguez drodrig...@fidelus.com
To: Girard, Jeffrey COL MIL USA; 'ccie_voice@onlinestudylist.com'
ccie_voice@onlinestudylist.com
Sent: Fri Oct 30 20:13:39 2009
Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 5.8 Testing failover to MGCP
Jeff - not really sure how the voice-port up/down status would affect the
SIP trunk from CUCMs perspective though. It should be purely IP end to end
unless you're hairpinning to TDM. Sorry I don't have the lab in front of me,
maybe there's something I'm missing in terms of configs or task
requirements, but good to hear you got it working.
- Original Message -
From: Girard, Jeffrey COL MIL USA jeffrey.gir...@us.army.mil
To: Daniel Rodriguez; ccie_voice@onlinestudylist.com
ccie_voice@onlinestudylist.com
Sent: Fri Oct 30 22:02:27 2009
Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 5.8 Testing failover to MGCP
Daniel -
SIP to MGCP
PG indicated to do a shut on the voice port - created as a result of the
T1. However this does not appear to work correctly as the CUCM still sees
the SIP trunk up.
My solution was to go in and point the SIP trunk to a bad IP address and
then reset the trunk. I then tested my long distance call again and it
failed over to MGCP as it should
Jeff
- Original Message -
From: Daniel Rodriguez drodrig...@fidelus.com
To: Girard, Jeffrey COL MIL USA; 'ccie_voice@onlinestudylist.com'
ccie_voice@onlinestudylist.com
Sent: Fri Oct 30 19:18:31 2009
Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 5.8 Testing failover to MGCP
Hey Jeff - It sounds like you're failing over from a SIP trunk to an MGCP
gateway, but I read that you're shutting a voice-port. Did you mean H323
gateway instead of SIP Trunk? Sorry but I don't have the lab manual in front
of me. If you meant H323 gateway to MGCP, make sure your service parameter
Stop Routing on User Busy and Stop Routing on Unallocated Number are set
to False. If you did mean SIP trunk to MGCP, I'm not sure where the
voice-port comes into play? Hope this helps.
- Dan
- Original Message -
From: ccie_voice-boun...@onlinestudylist.com
ccie_voice-boun...@onlinestudylist.com
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Sent: Fri Oct 30 21:06:31 2009
Subject: [OSL | CCIE_Voice] Vol 1 Lab 5.8 Testing failover to MGCP
Have completed the config and calls and transformations occur as they
should. To test the failover from HQ SIP GW to MGCP, I follow the
instructions in the PG and do a shut on the voice port on HQ GW. I retry the
call and get reorder tone. If a no shut the voice port and then go and
reverse the priority of the GWs in the RL (putting BR1 on top of HQ) and
then retry the call - it completes out through BR1 as it should with the
proper ANI. So, it does not appear that doing a shut on the HQ voice port is
the right way to test failover. Anybody else have this issue or have a
better way to test failover?
Jeff
___
For more information regarding industry leading CCIE Lab training, please visit
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Re: [OSL | CCIE_Voice] Vol 1 Lab 5.12. 4 digit dialing through GK -still not working
You should set the significant digits from the gateway configuration page, “I think you made it from the trunk configuration page” Thanks and best regards, EHAB SALEM Cisco Instructor | Sigma IT – Egypt LOGO6.png Suez Canal Tower, 70 El-Nil St., Level 14, 26, Dokki, Giza, Egypt Tel: +202 37482098 Fax: +202 37496536 Mobile: +2010 6224961 Email: esalem mailto:hsa...@sigma-it.net @sigma-it.net Website: http://www.sigma-it.net/ www.sigma-it.net -- This e-mail communication and any attachments thereto contain information which is confidential and are intended only for the use of the individuals or entities named above. If you are not the intended recipient, you are hereby notified that any disclosure, copying, distribution or the taking any action in reliance on the contents of these documents is strictly prohibited and may be illegal. Please notify us of your receipt of this e-mail in error and delete the e-mail and any copies of it From: Jeff Knuckle [mailto:jknuc...@nationwidelab.com] Sent: Thursday, November 05, 2009 8:52 PM To: Girard, Jeffrey COL MIL USA; aamir.panjw...@ivision.com.au; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 5.12. 4 digit dialing through GK -still not working Was the DNA tool able to provide you with any useful info? From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Girard, Jeffrey COL MIL USA Sent: Sunday, November 01, 2009 1:12 PM To: aamir.panjw...@ivision.com.au; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 5.12. 4 digit dialing through GK -still not working Update - I loaded up RTMT this morning and retested RTMT output indicates that the incoming call is not picking up the CSS-Internal RTMT line reads: DaReq.partitionSearchSpace(),filteredPartitionSearchSpaceString(),partitionSearchSpaceString() Followed several lines down by Potentialmatches=NoPotentialMatches Which is why I get the Call can't be completed as dialed I have verified/reverified that the H225 GK controlled trunk has CSS_Internal and that PT_Internal is in that CSS with 5002 using that PT To test my theory, I took 5001 out of the PT_Internal and placed it into the none partition. Called from 3002 to 5001 and it went through. Tried 3002 to 5002 and got the same error. I confirmed both calls were going thru GK with debug gatek main 10 So, any ideas why my trunk is not getting the internal partition? Jeff _ From: Aamir Panjwani aamir.panjw...@ivision.com.au To: Girard, Jeffrey COL MIL USA; ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Sat Oct 31 17:22:14 2009 Subject: RE: [OSL | CCIE_Voice] Vol 1 Lab 5.12. 4 digit dialing through GK Paste GK config section and output of “sh gatek gw” and “sh gatek end” From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Girard, Jeffrey COL MIL USA Sent: Sunday, 1 November 2009 10:32 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Vol 1 Lab 5.12. 4 digit dialing through GK Calls fromm HQ thru GK to BR2 work fine as well as from BR1 thru GK to BR2 Reverse clls from BR2 thu GK to CUCM gat Call can't be cpleted as dialed Debug voice dialpeer on BR2 shows correct DP is matching and adding texh-prefix of 1# Debug gatekeeper main 10 shows 1# tech prefix matches as well as zone prefix of 5 with remainder of 001 Source and destination zones both match to PL On CUCM, GK trunk has inbound calls sig digits set to 4 with CSS_Internal. No calling / called transformation CSSs set Ideas? Jeff __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ image001.png___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Volume 1 Lab5
Dear All, In Volume 1 Lab5, he's asking to implement call privileges between phones, and for internal and external calls. I would recommend to implement it using the CSS Line/Device approach: which is putting in the Device CSS a CSS which allows all calls, and in the line CSS a CSS which blocks the unwanted ones. What do you see? Thanks, Ehab M. Salem ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CME Ephone-dn registration with GK
Hi All,
I've configured my CME BR2-RTR to register with the Gatekeeper, I need the
BR2-RTR not to register its ephone-dns, so this is the configuration on the
BR2-RTR:
interface Loopback0
ip address 10.10.110.3 255.255.255.255
ip ospf network point-to-point
h323-gateway voip interface
h323-gateway voip id PL ipaddr 10.10.110.1 1719
h323-gateway voip h323-id BR2-RTR
h323-gateway voip tech-prefix 3
h323-gateway voip bind srcaddr 10.10.110.3
!
telephony-service
no auto-reg-ephone
max-ephones 4
max-dn 5 no-reg
ip source-address 10.10.110.3 port 2000
auto assign 1 to 2
network-locale ES
network-locale 1 ES
network-locale 2 ES
network-locale 3 ES
network-locale 4 ES
time-zone 28
time-format 24
date-format dd-mm-yy
max-conferences 8 gain -6
web admin system name admin password cisco
dn-webedit
transfer-system full-consult
create cnf-files version-stamp 7960 Nov 12 2009 10:31:52
!
!
ephone-dn 1 octo-line
number 3001 no-reg
description 32143001
name BR2-Phone 1
!
!
ephone 1
no phone-ui speeddial-fastdial
no phone-ui snr
no multicast-moh
device-security-mode none
mac-address 001C.58F0.7548
max-calls-per-button 5
busy-trigger-per-button 3
type 7970
button 1:1
and this is the Gatekeeper Configuration:
gatekeeper
zone local PL cisco.com 10.10.110.1
zone prefix PL 1... gw-priority 10 gk-trunk_1
zone prefix PL 1... gw-priority 9 gk-trunk_2
zone prefix PL 1... gw-priority 0 BR2-RTR
zone prefix PL 5... gw-priority 10 gk-trunk_1
zone prefix PL 5... gw-priority 9 gk-trunk_2
zone prefix PL 5... gw-priority 0 BR2-RTR
no shutdown
but it still registering the ephone-dn with the gatekeeper:
HQ-RTR#sh gatekeeper endpoints
GATEKEEPER ENDPOINT REGISTRATION
CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags
--- - --- - - -
10.10.110.3 1820 10.10.110.3 62279 PLH323-GW
H323-ID: BR2-RTR
E164-ID: 3001
Voice Capacity Max.= Avail.= Current.= 0
HQ-RTR#debug h225 asn1
value RasMessage ::= registrationRequest :
{
requestSeqNum 145
protocolIdentifier { 0 0 8 2250 0 4 }
discoveryComplete TRUE
callSignalAddress
{
ipAddress :
{
ip '0A0A6E03'H
port 1820
}
}
rasAddress
{
ipAddress :
{
ip '0A0A6E03'H
port 62279
}
}
terminalType
{
vendor
{
vendor
{
t35CountryCode 181
t35Extension 0
manufacturerCode 18
}
productId '436973636F47617465776179'H
versionId '32'H
}
gateway
{
protocol
{
voice :
{
supportedPrefixes
{
{
prefix dialedDigits : 3
}
}
},h323 :
{
supportedPrefixes
{
}
}
}
}
mc FALSE
undefinedNode FALSE
}
terminalAlias
{
h323-ID : {BR2-RTR},
dialedDigits : 3001
}
gatekeeperIdentifier {PL}
endpointVendor
{
vendor
{
t35CountryCode 181
t35Extension 0
manufacturerCode 18
}
productId '436973636F47617465776179'H
versionId '32'H
}
timeToLive 60
keepAlive FALSE
willSupplyUUIEs FALSE
maintainConnection TRUE
usageReportingCapability
{
nonStandardUsageTypes
{
{
nonStandardIdentifier h221NonStandard :
{
t35CountryCode 181
t35Extension 0
manufacturerCode 18
}
data '40'H
}
}
startTime NULL
endTime NULL
terminationCause NULL
}
}
Any idea?
Thanks,
Ehab M. Salem
___
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com
Re: [OSL | CCIE_Voice] CME Ephone-dn registration with GK
Thanks all for your reply... I made shut and no shut for the gatekeeper, and restarted the gateway from the BR2-RTRand didn't work also. But it WORKED NOW after reloading the BR2-RTR like what Hussam said Thanks guys... Ehab M. Salem -- This e-mail communication and any attachments thereto contain information which is confidential and are intended only for the use of the individuals or entities named above. If you are not the intended recipient, you are hereby notified that any disclosure, copying, distribution or the taking any action in reliance on the contents of these documents is strictly prohibited and may be illegal. Please notify us of your receipt of this e-mail in error and delete the e-mail and any copies of it -Original Message- From: hah...@sigma-it.net [mailto:hah...@sigma-it.net] Sent: Thursday, November 12, 2009 1:15 PM To: Ehab Salem Cc: ccie_voice@onlinestudylist.com Subject: Re: CME Ephone-dn registration with GK Dear Ehab, after configuring no-reg for all the ephone-dn you have to reload your CME. On Thu, 12 Nov 2009 12:39:52 +0200, Ehab Salem esa...@sigma-it.net wrote: Hi All, I've configured my CME BR2-RTR to register with the Gatekeeper, I need the BR2-RTR not to register its ephone-dns, so this is the configuration on the BR2-RTR: interface Loopback0 ip address 10.10.110.3 255.255.255.255 ip ospf network point-to-point h323-gateway voip interface h323-gateway voip id PL ipaddr 10.10.110.1 1719 h323-gateway voip h323-id BR2-RTR h323-gateway voip tech-prefix 3 h323-gateway voip bind srcaddr 10.10.110.3 ! telephony-service no auto-reg-ephone max-ephones 4 max-dn 5 no-reg ip source-address 10.10.110.3 port 2000 auto assign 1 to 2 network-locale ES network-locale 1 ES network-locale 2 ES network-locale 3 ES network-locale 4 ES time-zone 28 time-format 24 date-format dd-mm-yy max-conferences 8 gain -6 web admin system name admin password cisco dn-webedit transfer-system full-consult create cnf-files version-stamp 7960 Nov 12 2009 10:31:52 ! ! ephone-dn 1 octo-line number 3001 no-reg description 32143001 name BR2-Phone 1 ! ! ephone 1 no phone-ui speeddial-fastdial no phone-ui snr no multicast-moh device-security-mode none mac-address 001C.58F0.7548 max-calls-per-button 5 busy-trigger-per-button 3 type 7970 button 1:1 and this is the Gatekeeper Configuration: gatekeeper zone local PL cisco.com 10.10.110.1 zone prefix PL 1... gw-priority 10 gk-trunk_1 zone prefix PL 1... gw-priority 9 gk-trunk_2 zone prefix PL 1... gw-priority 0 BR2-RTR zone prefix PL 5... gw-priority 10 gk-trunk_1 zone prefix PL 5... gw-priority 9 gk-trunk_2 zone prefix PL 5... gw-priority 0 BR2-RTR no shutdown but it still registering the ephone-dn with the gatekeeper: HQ-RTR#sh gatekeeper endpoints GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags --- - --- - - - ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP Phone with CME
Guys, I'm facing a problem with my 7941 SIP Phone, it never ask for SIPDefault.cnf or SIPMac.cnf It's only asking for : TFTP: Looking for CTLSEP001C58F9F8EB.tlv TFTP: Looking for SEP001C58F9F8EB.cnf.xml TFTP: Looking for XMLDefault.cnf.xml This is my Configuration: tftp-server usbflash1:cmterm-7941_7961-sip.8-5-2SR1.cop.sgn alias cmefirm ! voice register global mode cme source-address 10.10.202.1 port 5060 max-dn 8 max-pool 2 load 7941 cmefirm create profile sync 0005133635525443 ! voice register dn 1 number 3005 ! voice register pool 1 id mac 001C.58F9.F8EB number 1 dn 1 dtmf-relay sip-notify any idea, Thanks, Ehab M. Salem ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] NTP Problem
Thanks Rebot, I think this is the problem because the time is different in each phone :-) Thanks, Ehab From: Rebot Gaber [mailto:rebotc...@hotmail.com] Sent: Monday, November 09, 2009 8:51 PM To: esa...@sigma-it.net; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] NTP Problem I had the same problem and I solved it by using vmware v 6.5 _ From: esa...@sigma-it.net To: ccie_voice@onlinestudylist.com Date: Mon, 9 Nov 2009 11:03:13 +0200 Subject: [OSL | CCIE_Voice] NTP Problem Hi All, I have a problem with the Time setup in my CUCM7.1 running on VMware I configured a VG as NTP server, and synchronized my CUCM with this VG (from the OS Admin page) Created Date/Time Group, assigned it to a Device pool and gave this Device pool to SCCP Phone But the time displayed on the phone is wrong!!! Although my SIP Phones are getting the correct time from the VG using NTP Reference. Any idea? Thanks. Ehab M. Salem _ Windows Live Hotmail: Your friends can get your Facebook updates, right from Hotmail http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/so cial-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_4:092009 R. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP Phone with CME
You are right, I found that I'm using the wrong Firmware file, I'm uploading the correct files now to the CME and try again. Thanks, Ehab M. Salem From: Rebot Gaber [mailto:rebotc...@hotmail.com] Sent: Tuesday, November 10, 2009 11:05 AM To: esa...@sigma-it.net Subject: RE: [OSL | CCIE_Voice] SIP Phone with CME I don't think this is the correct F/W you are using , because what I used is agroup of files not a single file, because SIP F/W for CUCM is different than CME SIP F/W _ From: esa...@sigma-it.net To: ccie_voice@onlinestudylist.com Date: Tue, 10 Nov 2009 10:08:58 +0200 Subject: Re: [OSL | CCIE_Voice] SIP Phone with CME Guys, I'm facing a problem with my 7941 SIP Phone, it never ask for SIPDefault.cnf or SIPMac.cnf It's only asking for : TFTP: Looking for CTLSEP001C58F9F8EB.tlv TFTP: Looking for SEP001C58F9F8EB.cnf.xml TFTP: Looking for XMLDefault.cnf.xml This is my Configuration: tftp-server usbflash1:cmterm-7941_7961-sip.8-5-2SR1.cop.sgn alias cmefirm ! voice register global mode cme source-address 10.10.202.1 port 5060 max-dn 8 max-pool 2 load 7941 cmefirm create profile sync 0005133635525443 ! voice register dn 1 number 3005 ! voice register pool 1 id mac 001C.58F9.F8EB number 1 dn 1 dtmf-relay sip-notify any idea, Thanks, Ehab M. Salem _ Windows Live Hotmail: Your friends can get your Facebook updates, right from Hotmail http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/so cial-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_4:092009 R. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP Phone with CME
Worked!!! Thanks. From: Ehab Salem [mailto:esa...@sigma-it.net] Sent: Tuesday, November 10, 2009 11:09 AM To: 'Rebot Gaber' Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SIP Phone with CME You are right, I found that I'm using the wrong Firmware file, I'm uploading the correct files now to the CME and try again. Thanks, Ehab M. Salem From: Rebot Gaber [mailto:rebotc...@hotmail.com] Sent: Tuesday, November 10, 2009 11:05 AM To: esa...@sigma-it.net Subject: RE: [OSL | CCIE_Voice] SIP Phone with CME I don't think this is the correct F/W you are using , because what I used is agroup of files not a single file, because SIP F/W for CUCM is different than CME SIP F/W _ From: esa...@sigma-it.net To: ccie_voice@onlinestudylist.com Date: Tue, 10 Nov 2009 10:08:58 +0200 Subject: Re: [OSL | CCIE_Voice] SIP Phone with CME Guys, I'm facing a problem with my 7941 SIP Phone, it never ask for SIPDefault.cnf or SIPMac.cnf It's only asking for : TFTP: Looking for CTLSEP001C58F9F8EB.tlv TFTP: Looking for SEP001C58F9F8EB.cnf.xml TFTP: Looking for XMLDefault.cnf.xml This is my Configuration: tftp-server usbflash1:cmterm-7941_7961-sip.8-5-2SR1.cop.sgn alias cmefirm ! voice register global mode cme source-address 10.10.202.1 port 5060 max-dn 8 max-pool 2 load 7941 cmefirm create profile sync 0005133635525443 ! voice register dn 1 number 3005 ! voice register pool 1 id mac 001C.58F9.F8EB number 1 dn 1 dtmf-relay sip-notify any idea, Thanks, Ehab M. Salem _ Windows Live Hotmail: Your friends can get your Facebook updates, right from Hotmail http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/so cial-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_4:092009 R. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Conference in CME SIP Phone
Hi All, I'm trying to disable the conference softkey for a SIP Phone on CME, this is the configuration: voice register template 1 no conference enable ! Voice register pool 1 Template 1 ! Voice register global Create profile Reset But still I can make conference from this phone.the only way was I removed the conference softkey from the phone template and it worked. Any idea. Thanks Ehab M. Salem ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] NTP Problem
Hi All, I have a problem with the Time setup in my CUCM7.1 running on VMware I configured a VG as NTP server, and synchronized my CUCM with this VG (from the OS Admin page) Created Date/Time Group, assigned it to a Device pool and gave this Device pool to SCCP Phone But the time displayed on the phone is wrong!!! Although my SIP Phones are getting the correct time from the VG using NTP Reference. Any idea? Thanks. Ehab M. Salem ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
