Re: [OSL | CCIE_Voice] no DTMF Relay to UC via PSTN during HA (testing AAR)
Do you have the command Mgcp dtmf codec all out In your mgcp config From: Vignesh Sethuraman sethuvign...@gmail.commailto:sethuvign...@gmail.com Date: Tuesday, January 21, 2014 at 1:51 PM To: ccievoice ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] no DTMF Relay to UC via PSTN during HA (testing AAR) Hello All, Unity Connection not recognizing the password (no DTMF) when the call is routed as following during a high availability situation. SiteB PH2/PH3 --- MGCP T1 Port of SiteB GW My PSTN GW (use to switch call between all sites via pots dialpeers) - SiteA H323 GW - CUCM SUB Unity Connection. * The Unity Connection is playing Message -- Enter you PIN * Unity Connection recognizes SiteB PH2 is a registered user's number , so asks for password * When pressing password unity connection does not recognize that any key is pressed I am facing the same issue as mentioned in the below link but I am using Skinny integration of CUC to CUCM. http://onlinestudylist.com/archives/ccie_voice/2013-August/085101.html Please let me know what I am missing. Thanks, Viki ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] Translation-rule help
Why not just make the pattern 9T? From: Regis Reis regis_r...@yahoo.com.brmailto:regis_r...@yahoo.com.br Reply-To: Regis Reis regis_r...@yahoo.com.brmailto:regis_r...@yahoo.com.br Date: Friday, June 28, 2013 12:02 PM To: Hesham Abdelkereem heshamcentr...@gmail.commailto:heshamcentr...@gmail.com, ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Translation-rule help Hi Hesham, You make this form: voice translation-rule 1 rule 1 /^91\(..$\)/ /\1/ rule 2 /^9\(..$\)/ /\1/ rule 3 /^9\(...$\)/ /\1/ Test it. I put the $ after last digit, because I understand that you want match with the total digits diled. Regis Reis De: Hesham Abdelkereem heshamcentr...@gmail.commailto:heshamcentr...@gmail.com Para: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Enviadas: Sexta-feira, 28 de Junho de 2013 13:29 Assunto: [OSL | CCIE_Voice] Translation-rule help Dear All, I would like to make a translation-rule to do the following remove 9 from 91[10 digits] remove 9 from 9[10 digits] remove 9 from 9[7 digits] i did it the following but was invalid voice translation-rule 1 rule 1 /^91../ /../ rule 2 /^9../ /../ rule 3 /^9.../ /.../ when i did it like that it didn't work I would like to make it strict match not like /^9/ // this will overlap Please help me whats the other way to do it. Thanks, Hesham ___ 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] keyboard model in SJ and RTP exam
Logitec 120 if I remember right, I saw them at Tiger Direct for $16. From: Tian id21...@gmail.commailto:id21...@gmail.com Date: Thursday, June 13, 2013 2:16 PM To: 'Karen Johnson' karen.johnson...@yahoo.camailto:karen.johnson...@yahoo.ca, 'nielsenj' niels...@gmail.commailto:niels...@gmail.com Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] keyboard model in SJ and RTP exam It’s Logitech, both are for keyboard and mouse. From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Karen Johnson Sent: Thursday, June 13, 2013 10:41 AM To: nielsenj Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] keyboard model in SJ and RTP exam hmm, i remembered only see cheap keyboard, seem like Dell. but forgot the model From: nielsenj niels...@gmail.commailto:niels...@gmail.com To: Karen Johnson karen.johnson...@yahoo.camailto:karen.johnson...@yahoo.ca Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Sent: Tuesday, May 28, 2013 7:45:49 PM Subject: Re: [OSL | CCIE_Voice] keyboard model in SJ and RTP exam Logitech K120 On Tue, May 28, 2013 at 4:36 PM, Karen Johnson karen.johnson...@yahoo.camailto:karen.johnson...@yahoo.ca wrote: hi folks, anyone remember what is the keyboard model use in SJ and RTP, need to duplicate for speed. tks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] !!Questions on H323 trunk troubleshooting!!
Just configure it as an h323 gateway, it is still a trunk and it is the shortest distance between two points. From: Robert Thomas tho...@gmail.commailto:tho...@gmail.com Date: Tuesday, April 30, 2013 9:00 PM To: ikizoo hello ikiz...@hotmail.commailto:ikiz...@hotmail.com Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com, sanity insanity networksanitytoinsan...@gmail.commailto:networksanitytoinsan...@gmail.com Subject: Re: [OSL | CCIE_Voice] !!Questions on H323 trunk troubleshooting!! I don't think you go for the ICT of the H323 Gateway. The recommendation is a H225 Trunk. When you create a new trunk you have multiple options: ICT GK controlled, ICT non GK controlled and H255 Trunk. I would go for the last. On Thu, Apr 25, 2013 at 12:55 PM, ikizoo hello ikiz...@hotmail.commailto:ikiz...@hotmail.com wrote: i have different perspective,, according to cisco recommendation ICT is only for between 2 CUCM clusters, and lab requirement saying POC of H323 trunk for PSTN access. so i would choose GW instead. -ikizoo From: mmac...@bcbsal.orgmailto:mmac...@bcbsal.org To: networksanitytoinsan...@gmail.commailto:networksanitytoinsan...@gmail.com; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Date: Thu, 25 Apr 2013 08:23:21 -0500 Subject: Re: [OSL | CCIE_Voice] !!Questions on H323 trunk troubleshooting!! 1)The question says trunk so it must be a trunk (inter cluster trunk) 2) that’s more difficult. Usually it’s a codec mismatch. And is found in the trace ( I forget exact wording you are looking for). Hopefully someone else can provide better detail. M From:ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of sanity insanity Sent: Thursday, April 25, 2013 12:00 AM To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] !!Questions on H323 trunk troubleshooting!! hi All, I have a few questions in relation to H323 troubleshooting We are required to provision a proof of concept (POC) H323 trunk between CUCM and voip service provider . The cucm should send H323 traffic to service provider on ip address XX.XX.XX.XX (public Ip address) . This call should be available for HQ Phones only. Inbound H.323 calls from VoIP service provider is not needed for the PoC test. Questions :- 1) Should we use a H323 gateway or H323 trunk ( no gatekeeper controlled) to achieve the above requirement? 2) If we need provide the appropriate debug or traces to verify whether or not No way audio instances prevalent for HQ Phones . Then what do we look for in the logs ? Thanks , MJ ***CONFIDENTIALITY NOTICE*** This e-mail is intended for the sole use of the individual(s) to whom it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. You are hereby notified that any dissemination, duplication, or distribution of this transmission by someone other than the intended addressee or its designated agent is strictly prohibited. If you receive this e-mail in error, please notify me immediately by replying to this e-mail. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.PlatinumPlacement.com -- Robert Thomas Zamora tho...@gmail.commailto:tho...@gmail.com +50689389544 http://cr.linkedin.com/pub/robert-thomas-zamora/29/913/8a8 CCNP, CCNP Voice ___ 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] WAN QoS Calculations
From those numbers, those are the CIR values of the PVC, so you have to use those values. They physical circuit would be a T1, which is academic as the carrier can discard above the committed rate. Hope that helps, Mark From: Barrera, Hugo hugo.barr...@nexusis.commailto:hugo.barr...@nexusis.com Date: Tuesday, April 9, 2013 4:09 PM To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] WAN QoS Calculations QoS Guru’s, In the real lab I know I have to do some calculations utilizing 95% of the bandwidth…so if there is a link between SA and SB of 384k and SA and SC of 768k is the 95% from these numbers or what the actual interface can do? Also what is a simple straight to the point read on this, I really don’t want to review an srnd? Hugo ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Lab 1: Task 3.1 : 5 Lab Handbook - workbook lab
It is the default, so it will not show. You only need the command if you are emulating the network. From: Ramcharan Arya ramcharan.a...@gmail.commailto:ramcharan.a...@gmail.com Date: Thursday, April 4, 2013 9:54 AM To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Lab 1: Task 3.1 : 5 Lab Handbook - workbook lab Hi, I have question regarding User-side Q.921/Q.931 Signalling when I configure my gateway will full T1 PRI command isdn protocol-emulate user does not appear under D channel configuration. interface Serial0/0/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn map address 911 plan isdn type unknown isdn map address [2-9]..$ plan isdn type subscriber isdn map address 1[2-9]..[2-9]..$ plan isdn type national isdn outgoing display-ie isdn outgoing ie redirecting-number no cdp enable Do we need to add this command manual.? WB says it is by default. Please let us know your thoughts Thanks Regards, Ramcharan Arya ___ 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] H323 One Way Audio Troubleshooting
MGCP gateways show that in the closing message with packets sent and received (debug mgcp packet). Not sure how you would do that with H323. __ Mark Thrash marth...@cisco.commailto:marth...@cisco.com NCE, Collaboration and Unified Communications Practice MSTM, MCSE, CCIE R/S 2405 office 408-894-2086 mobile 918-671-3237 __ The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer. __ From: CISCO CCIE VOICE ccievoic...@gmail.commailto:ccievoic...@gmail.com Date: Tuesday, March 19, 2013 3:19 AM To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] H323 One Way Audio Troubleshooting Hi Experts, Can any one share there knowledge and experience on how to troubleshoot one-way audio when the call is answer from PSTN phone which messages do i need to look at on RTMT and which traces do i need to enable on CUCM to check the One way audio problem .. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] H323 One Way Audio Troubleshooting
Wireshark! I forgot about that! __ Mark Thrash marth...@cisco.commailto:marth...@cisco.com NCE, Collaboration and Unified Communications Practice MSTM, MCSE, CCIE R/S 2405 office 408-894-2086 mobile 918-671-3237 __ The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer. __ From: Cisco Employee marth...@cisco.commailto:marth...@cisco.com Date: Wednesday, March 20, 2013 1:39 AM To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] H323 One Way Audio Troubleshooting MGCP gateways show that in the closing message with packets sent and received (debug mgcp packet). Not sure how you would do that with H323. __ Mark Thrash marth...@cisco.commailto:marth...@cisco.com NCE, Collaboration and Unified Communications Practice MSTM, MCSE, CCIE R/S 2405 office 408-894-2086 mobile 918-671-3237 __ The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer. __ From: CISCO CCIE VOICE ccievoic...@gmail.commailto:ccievoic...@gmail.com Date: Tuesday, March 19, 2013 3:19 AM To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] H323 One Way Audio Troubleshooting Hi Experts, Can any one share there knowledge and experience on how to troubleshoot one-way audio when the call is answer from PSTN phone which messages do i need to look at on RTMT and which traces do i need to enable on CUCM to check the One way audio problem .. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] HQ and Branch1 phones cannot call CUE
CUE only does G711, do you have a transcoder? Sent from my iPhone On Mar 20, 2013, at 12:39 PM, Vikky Kumar vikkyne...@gmail.com wrote: Hi Experts, I configured branch 2 CME/CUE working normal for Voice mails. CUE is registered with CUCM but I call not call CUE(6220) from HQ and Branch1 but branch2 phones 4001 and 4002 working normal/ringing from every where. FYI. I have also configured CAC between on Br2 site - HQ site Please hel. Regards Vikky ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension
What about a calling party transform mask on the incoming gateway? Sent from my iPhone On Mar 20, 2013, at 10:43 PM, Steve Keller skeller...@gmail.commailto:skeller...@gmail.com wrote: Thanks Bill, I like this option pretty well as it seems to limit treatment of calls this way to CUC when site B is in SRST mode only. I will try to lab this up tomorrow morning. Question for you, will this only solve my issue of pressing the VM button to access my mailbox to retrieve a message. Meaning when PSTN calls in to site B phone and then gets forward(redirected) to voicemail, I use a dial-peer that provides RDNIS capabilites to route the caller to the correct mailbox and not the opening greeting. So with this would i still want to use the following to get the caller into my mailbox? dial-peer voice 2600 pots destination-pattern 2600 port 0/0/0:23 no digit-strip prefix 202555 ( assuming no LD code at this site ) this is the way i get callers into my mailbox - using RDNIS. On Wed, Mar 20, 2013 at 10:49 PM, William Bell b...@ucguerrilla.commailto:b...@ucguerrilla.com wrote: If you are told National calls must present a 10D ANI AND you are restricted from using an alternate extension in CUC then I do the following. I am not sure whether this would be graded right or wrong On the SRST device (assume basic SRST) call-manager-fallback max-ephone 10 max-dn 20 oct huntst chan 1 voicemail 912025552699 ! or some unused DID on Site A call-forward noan 912025552600 time 20 !assume VM pilot is 2600 call-forward busy 912025552600 time-z whatever time-f whatever date-f whatever call-forward pattern .T ! do your dial-peer work, as needed On CUCM: Create a PT: hq_gw-in_pt Create a CSS: hq_gw_css Assign CSS to hq gateway Either a.) create a translation in hq_gw-in_pt Pattern: 2699 xform ANI: xform DNIS: 2600! as in, redirect to regular VM pilot CSS: your regular HQ phone CSS will do OR b.) create a new hunt pilot in hq_gw-in_pt Pattern: 2699 HL: your VM HL xform ANI: Why would I go this path? 1. We had a requirement that National calls are presented with a 10D ANI in SRST mode. I assume that you would already have a translation-p that handles this bit 2. We can't modify the CUC subscriber. 3. This method doesn't interfere with RDNIS to VM 4. This method doesn't interfere with direct or redirect calls from HQ or SiteC Anyway, that is my 2 cents. -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 20, 2013, at 9:33 PM, Bill wrote: Traditionally you would use the alternate extension or a on the pilot. So if you we're denied the ability to use alternate extension for this task but had to use it for another, say allowing easy voicemail access to a user at home, then I think you are looking at a very specific inbound translation on your gateway or nay sending 4 digits if the PSTN allows. I would definitely test out the translation setup to ensure you can do it. Sent from my iPad On Mar 20, 2013, at 3:44 PM, Steve Keller skeller...@gmail.commailto:skeller...@gmail.com wrote: In SRST mode, when the vm button is pressed, i have a dial-peer to route this call to the vm hunt pilot on the UCM. dial-peer voice 2600 pots description voicemail-pilot destination-pattern 2600 no digit-strip port 0/0/0:23 prefix 1408202 If i have to adhere to the requirement that LD calls should be 10 digit ANI, then i am sending the full 10 digit ANI for this call as well ( even though it more of a hidden number rather than an implicit user dialed number) Thus the call arrives at site A GW with 10 digits , say 9723033002tel:9723033002. In order to route this call to the correct mailbox i would have to use Alternate Extension of 9723033002tel:9723033002 and then i will be prompted to login. However, if i am not allowed to use alternate extension then i must have another strategy. here are the choices i can think of, please chime in if you too have experienced this dilemma and what is the best way to solve it. 1) do not send the full 10 digit ANI for this call and it will arrive at site a GW as 4 digit ANI and then land in the mailbox, but not adhere to LD calls should be 10 digit ANI requirement. 2) put as calling party transform mask on the Hunt Pilot, thus stripping the caller ANI to 4 digits and i can be prompted to log in. However i think with this method, anytime the caller ANI is read to before the message is played the caller id would incorrectly state from 3002 instead of from 9723033002tel:9723033002 essentially, what is the best way for SRST users to access voicemail when you are not permitted to use Alternate Extension. thanks in advance all!! steve ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out
Re: [OSL | CCIE_Voice] MVA Mobile can't call Associated Desk Number
You need more dn's. can't you use a back-to-back PRI with say 4 channels? Otherwise I think your results are misleading. I may not understand you however, if that's the case, maybe someone else will chime-in. Sorry - good luck Sent from my iPhone On Mar 18, 2013, at 10:36 PM, Suresh Bhandari bring...@gmail.commailto:bring...@gmail.com wrote: This is because I have only local PSTN DN on, say site B, to which my internal DN is associated using MVA. Now if the requirement is that the internal number should be able to cal the local number of that site, and due to limited resource, we don't have another local PSTN in the said site. What will be your solution in this case? Thanks. On Tue, Mar 19, 2013 at 10:47 AM, Mark Thrash (marthras) marth...@cisco.commailto:marth...@cisco.com wrote: I'm confused; why would you call your desk phone from your cell phone? You should be able to call another DN on the cluster and see the from CLID as being from your desk phone and the in-use LED on your phone should be red. Sent from my iPhone On Mar 18, 2013, at 10:00 AM, Suresh Bhandari bring...@gmail.commailto:bring...@gmail.com wrote: Experts! Had been working on MVA. Just to drop a question on RDP association. When we associate the desk phone DN with RDP, both DN's must be in the same partition, so that users can see remote in use on the desk phone and pickup the call within configured time from the desk phone. What I experienced that, when I try to call the internal DN, using the RDP associated PSTN number, the call is failing, with Temporary Failure in isdn debugs. is this an expected behavior? What if the question requirement is to get a call from same PSTN to the associated internal DN? I know we can separate the DNs putting them in separate partitions, but, then desk phone pickup is not happening. Any ideas? TIA -- Suresh Bhandari ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.PlatinumPlacement.com -- Suresh Bhandari ___ 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] MVA Mobile can't call Associated Desk Number
I'm confused; why would you call your desk phone from your cell phone? You should be able to call another DN on the cluster and see the from CLID as being from your desk phone and the in-use LED on your phone should be red. Sent from my iPhone On Mar 18, 2013, at 10:00 AM, Suresh Bhandari bring...@gmail.com wrote: Experts! Had been working on MVA. Just to drop a question on RDP association. When we associate the desk phone DN with RDP, both DN's must be in the same partition, so that users can see remote in use on the desk phone and pickup the call within configured time from the desk phone. What I experienced that, when I try to call the internal DN, using the RDP associated PSTN number, the call is failing, with Temporary Failure in isdn debugs. is this an expected behavior? What if the question requirement is to get a call from same PSTN to the associated internal DN? I know we can separate the DNs putting them in separate partitions, but, then desk phone pickup is not happening. Any ideas? TIA -- Suresh Bhandari ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] isdn outgoing ie facility
I use ISDN OUTGOING DISPLAY-IE to push the caller's name out the PSTN. That should work in the lab From: donny f f.faraday...@gmail.commailto:f.faraday...@gmail.com Date: Wednesday, March 13, 2013 11:33 PM To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] isdn outgoing ie facility hi all, does anybody can help me understand what this command for and when we need to use ? -isdn outgoing ie facility tks -- Original message -- From:Sergey Heyphets ser...@heyphets.commailto:ser...@heyphets.com Date: 13 Mar 13 18:38:09 Subject: Re: [OSL | CCIE_Voice] Navigating to the link. To: Cc: Online Study (ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com) ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Products -- Voice and Unified Communications -- IP Telephony -- Unified Communications Endpoints -- Cisco Unified IP Phone 7900 Series -- Maintain and Operate -- Maintain and Operate Guides -- Cisco Unified IP Phone 7965G and 7945G Administration Guide for Cisco Unified Communications Manager 6.1 -- Customizing the Cisco Unified IP Phone On Wed, Mar 13, 2013 at 6:33 AM, singh singh8...@in.commailto:singh8...@in.com wrote: Hi Guys, I am trying to navigate to this link: - http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7965g_7945g/6_1/english/administration/guide/7965cst.html from Here : http://www.cisco.com/cisco/web/psa/default.html Do anyone know the navigation path to get to the 7965/7945 guide above with the default link? -singh Dear ccie_voice! Get Yourself a cool, short @in.com Email ID now!http://www3.in.com/sso/commonregister.php?ref=INutm_source=inviteutm_medium=outgoing ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.platinumplacement.com/ Get Yourself a cool, short @in.com Email ID now!http://www3.in.com/sso/commonregister.php?ref=INutm_source=inviteutm_medium=outgoing ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://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] Using BAT in LAB
Oh yea __ Mark Thrash marth...@cisco.commailto:marth...@cisco.com NCE, Collaboration and Unified Communications Practice MSTM, MCSE, CCIE R/S 2405 office 408-894-2086 mobile 918-671-3237 __ The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer. __ From: CISCO CCIE VOICE ccievoic...@gmail.commailto:ccievoic...@gmail.com Date: Tuesday, February 26, 2013 2:44 PM To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Using BAT in LAB Hi, Is it safe to use BAT in real lab exam? thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Hyphothetical Voice Infrastructure Issue Question
Verified CDP ver 2 running? __ Mark Thrash marth...@cisco.commailto:marth...@cisco.com NCE, Collaboration and Unified Communications Practice MSTM, MCSE, CCIE R/S 2405 office 408-894-2086 mobile 918-671-3237 __ The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer. __ From: Michael Davis michaeldavis1...@yahoo.commailto:michaeldavis1...@yahoo.com Reply-To: Michael Davis michaeldavis1...@yahoo.commailto:michaeldavis1...@yahoo.com Date: Thursday, December 13, 2012 6:38 AM To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Hyphothetical Voice Infrastructure Issue Question Scenario for campus intrasture problem: Phones are not registering. Phones are obtaining a valid IP address and have been erased to ensure they are getting a fresh IP address. Solutions investigated Verified VLAN's are correct and active Verified option 150 is correct in the DHCP options Verified Helper IP address is on was the voice vlan's when in different subnet TFTP and DHCP on CM was restarted. NTP was verified and CM DB replication is normal. Idea that not checkd: WAN Qos may have been enabled thereby causing phone registration issues. Am I on the right track? Michael Davis ___ 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] Hyphothetical Voice Infrastructure Issue Question
Vlan pruned on an interface someplace? __ Mark Thrash marth...@cisco.commailto:marth...@cisco.com NCE, Collaboration and Unified Communications Practice MSTM, MCSE, CCIE R/S 2405 office 408-894-2086 mobile 918-671-3237 __ The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer. __ From: Michael Davis michaeldavis1...@yahoo.commailto:michaeldavis1...@yahoo.com Reply-To: Michael Davis michaeldavis1...@yahoo.commailto:michaeldavis1...@yahoo.com Date: Thursday, December 13, 2012 6:38 AM To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Hyphothetical Voice Infrastructure Issue Question Scenario for campus intrasture problem: Phones are not registering. Phones are obtaining a valid IP address and have been erased to ensure they are getting a fresh IP address. Solutions investigated Verified VLAN's are correct and active Verified option 150 is correct in the DHCP options Verified Helper IP address is on was the voice vlan's when in different subnet TFTP and DHCP on CM was restarted. NTP was verified and CM DB replication is normal. Idea that not checkd: WAN Qos may have been enabled thereby causing phone registration issues. Am I on the right track? Michael Davis ___ 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
