Re: [OSL | CCIE_Voice] 911 calls by mistake
We had the same problem. Unfortunately it's a user training issue. We decided to take out the 911 pattern and put stickers on the phones which specified to dial 9911 for emergency. False 911 calls dramatically reduced after that. HTH On May 26, 2014, at 8:31 AM, Ben John benjoh...@hotmail.com wrote: Guys, Our users are dialing 911 by mistake and cops are responding to the calls i think the reason being we use 9 to dial out. For long distance it is 91 and international it is 9011 . i thought about using 8 instead of 9 to dial out but we have some DNs that start with 8. Some of the guys suggest to use secondary dial tone when we press 9. Below are the route patterns that start with 9. Any idea how to solve this ? 9.011! 9.011!# 9.1[2-9]XX[2-9]XX 9.[2-9]XX[2-9]XX 91.[2-9]XX[2-9]XX 9.911 911 91.800[2-9]XX Thanks, Ben ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] Collaboration Lab 1
First of all, mind yr manners on the forum before you say anything outlandish on different races. Secondly using the website at the first place to get lab dumps shows yr character on how you approach CCIE. On Mar 17, 2014, at 10:20 AM, kit yee kity...@outlook.com wrote: Yes, ccielabdumps some indian cheater its just waste of time and fake i have asked my card company to reverse my funds. ccielabdumps is fake I request OSL to block his ip address and this name so that we will not see any such spam From: nexusg...@hotmail.com To: collabg...@gmail.com; ccie_voice@onlinestudylist.com; ccie_voice-requ...@onlinestudylist.com Date: Mon, 17 Mar 2014 09:49:32 -0400 Subject: Re: [OSL | CCIE_Voice] Collaboration Lab 1 Hi Collabguru, Yes ccielabdumps is fake website better don't use the fake stuff.. Below is the orginal link http://collaborationie.com/index.php?/forum/21-ccie-collaboration-lab/ Cheers Date: Mon, 17 Mar 2014 14:33:51 +0530 From: collabg...@gmail.com To: ccie_voice@onlinestudylist.com; ccie_voice-requ...@onlinestudylist.com Subject: [OSL | CCIE_Voice] Collaboration Lab 1 Hey Mates, Today I got lab 1 and what I got in the lab was exactly the same topology from here.Also the questions given in this book were similar to what I had in my exam http://www.4shared.com/office/t-pZpEZEba/CCIELABDUMPS_Collaboration_Lab.html Thanx Collabguru ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
[OSL | CCIE_Voice] AAR Configuration
Hi Guys, i am trying to configure AAR between 2 sites, when i assign AAR-GROUP and AAR-CSS on device pool it does not take effect rather i have to apply it each phone device and GW inorder for it to work.Is there any thing i am missing 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] AAR Configuration
Hi Justin, Thanks for your reply i will try that and wll update you on ths... Thanks On Mon, Nov 4, 2013 at 4:54 PM, Justin Carney justin.s.car...@gmail.comwrote: The aar-group setting on device pool does NOT get pushed to all devices in the device pool, while the aar-css does. My strategy is to set aar-css at the dev pool and to manually set the aar-group on EVERY device (phone, gw, vm or cti port, etc) and EVERY line/dn. This take no thought while provisioning thing and when I get to an aar question the only thing to build is the rlist (maybe, if an existing doesnt match exactly) and route pattern. That said my strategy is slightly overprovisioned to save time. I did thorough testing and came up with the minimum config for aar: 1. The calling entity must have the AAR-CSS on the DEVICE (there is no aar css field on a dn, it only exists on a device/port/gw) 2. The calling entity must have the AAR-GROUP set on Either Device *OR* Line/DN 3. The called/target LINE/DN must have the AAR-GROUP. (this makes sense, as you call a dn and you don't care which device(s) have a line appearance for this dn.) if the called DN doesn't have the aar-group it will NOT work, regarless of whether the device where the dn is assigned has the aar-group In summary, my strategy pit the group every where and the css on devices and I don't have to memorize the minimum req in the lab - or more importantly I don't revisit config pahes just to setup aar. Hope this helps... -Justin On Nov 4, 2013 6:57 AM, CISCO CCIE VOICE ccievoic...@gmail.com wrote: Hi Guys, i am trying to configure AAR between 2 sites, when i assign AAR-GROUP and AAR-CSS on device pool it does not take effect rather i have to apply it each phone device and GW inorder for it to work.Is there any thing i am missing Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Generate a report for number of calls into PRI
CDR/CAR should be able to provide breakdown by PRI since it's MGCP. On Sep 4, 2013, at 5:34 PM, Edgar Feliz ejzi...@gmail.com wrote: TELCO can provide a usage report for each PRI, who is the SP? Edgar On Tue, Sep 3, 2013 at 2:23 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Dear Experts, I have 12 PRI configured as MGCP gateways and would like to replace them by a CUBE. Now, I would like to make Statistics/Feasability study about the number of concurrent calls on each PRI for example today from 8am to 5PM. Is there is anyway I can do that? That will help me in the calculation to order the number of concurrent calls properly when I migrate into SIP. 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 ___ 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] Is the CCIE voice worth anymore?
+1, although large scale will be something like 20k+ phones. Many small and mid-size companies will move to cloud eventually, given the cost savings with infrastructure/IT. Office 365 is an example. On Aug 29, 2013, at 3:17 PM, Michael Davis michaeldavis1...@yahoo.com wrote: No matter what, there will ALWAYS been a need for large scale Enterprise voice systems. I am one of those people, and I am sure I am not alone, I will always want a physical phone. I am also one of these engineers who will always recommned a system that is directly under your own site's controll. Clouds are great, but they have their place. I don't think telecom will ever be a total cloud based solution. From: Bill Lake whl...@gmail.com To: Drake J jdrake...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Thursday, August 29, 2013 8:12 AM Subject: Re: [OSL | CCIE_Voice] Is the CCIE voice worth anymore? As a former big Telco employee, they want three things: Stability Scalability Profitability At this time these applications are not there. On Thu, Aug 29, 2013 at 6:45 AM, Bill Lake whl...@gmail.com wrote: As a former big Telco employee, they want three things: Stability Scalability On Thu, Aug 29, 2013 at 6:30 AM, Drake J jdrake...@gmail.com wrote: hi Laksh, Thanks for your inputs here.This was a good discussion. It is always good for us to all know about things that happen outside . Talking about Telco OTTs we can already see few of the Telcos have come out with Webrtc solutions for enterprise and service providers . Check this video out too depicting their solution... http://www.youtube.com/watch?v=Nz-BQZMp3sk Most of these applications written on software are supposed to open source and left for the users to customize . No real networking staff expertise required just download the SDK/API and customize and no more complex network topologies in future. Also no licensing fee too . Hence a real killer of techology in the future most likely we will see a wide spread of this starting 2014 if all predictions are to be believed. Hope someone from any of the TELCOs on this alias can add a few comments as well. Thanks once again for your inputs everyone. On Wed, Aug 28, 2013 at 11:05 PM, Lakshmish NS lakshmish...@gmail.com wrote: Hi Drake, I totally understand your concern, I'd be worried too. Having said that, we should always update ourselves with the latest technology. However, in future I believe Asterisk might be able to give tough run to Cisco UC. Not sure though, I hear stories that it is unstable and featureless compared to CUCM. I hope if someone aware of Asterisk would help us out here. Regard, Laksh On Wed, Aug 28, 2013 at 9:56 PM, Drake J jdrake...@gmail.com wrote: Hi Guys, Thanks for your responses I see u guys have empathized on call routing and and UC hardware for backend deployments. However Telco OTTs are coming up with directly provide these services over the cloud . Here is a disruptive analysis : http://www.slideshare.net/deanb/disruptive-analysis-web-rtc-overview-april-2013 Anyways, this might be not be so serious afterall . Just thought of brainstorming . Thanks guys for your responses again. On Tue, Aug 27, 2013 at 6:20 PM, Lakshmish NS lakshmish...@gmail.com wrote: Didn't have time to go through the video, I believe WebRTC is nothing but a Protocol, similar to SIP, H.323. Moreover, this protocol would only appeal to the Web audience, just like Skype, or Google talk. You still need to use UC hardware and their design for enterprise deployments. I mean we don't use Google talk and Skype in companies right? SIP is open source, but still Cisco uses it. As FAQ's suggest WebRTC is an open framework for the web that enables Real Time Communications in the browser. If only UC was that easy that could be implemented through browser, we didn't have to work this hard for CCIE numbers. You might want to go through this... http://www.webrtc.org/faq You've clearly misinterpreted WebRTC here.. On Tue, Aug 27, 2013 at 5:17 PM, Drake J jdrake...@gmail.com wrote: hi All, Had a troubling question hence thought of putting it out .Looking at the UC and networking trends worldwide it appears that the future of UC and collaboration is web based. Webrtc is the protocol that the world will use and individuals and organizations just need to code their requirement based on the WEBRTC. Here is the presentation that Google recently made http://www.youtube.com/watch?v=E8C8ouiXHHk Clearly many of the UC vendors are already losing out and will be losing out in year 2014. Most of the customers are already looking at reducing the cost involved in maintaining costly UC vendor networks and their networking staff . Therefore that brings
[OSL | CCIE_Voice] max calls and busy-trigger using ephone-template
Hi Experts, I am not able to restrict max-call-per-button 2 and busy-trigger-per-button 1 using ephone-template ephone-dn-template 1 call-forward busy 2220 call-forward noan 2220 timeout 20 huntstop channel 1 ephone-template 1 softkeys remote-in-use Newcall CBarge max-calls-per-button 2 busy-trigger-per-button 1 SiteB-RTR(config)#do sh run | s telep telephony-service sdspfarm units 2 sdspfarm tag 1 SB-CONF srst mode auto-provision none srst ephone template 1 srst dn template 1 srst dn line-mode octo max-ephones 10 max-dn 10 no-reg ip source-address 177.2.11.1 port 2000 timeouts interdigit 3 system message fallback mode time-zone 8 time-format 24 voicemail 2220 max-conferences 8 gain -6 call-forward pattern .T transfer-system full-consult transfer-pattern .T 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] HWIC-4ESW
ESW are around $200, while a POE switch is $50.. Figure that.. :) On Jul 26, 2013, at 10:20 AM, wilson.sam...@bt.com wrote: One could easily buy the HWIC-4ESW on the ebay, and its almost at the same price as a 3560-PoE, however with ESW you will have to use the power bricks for the phones or buy a POE - Daughter Card for the phones to get power, which will put them in a slightly expensive mode than a PoE Switch (sounds bit crazy, isn't it?) Regards -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of CCIE Voice Aspirant Sent: Thursday, July 25, 2013 12:45 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] HWIC-4ESW Hello list I am looking for 2 HWIC-4ESW cards for my lab, does anyone have spares I can buy? Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] HWIC-4ESW
Hello list I am looking for 2 HWIC-4ESW cards for my lab, does anyone have spares I can buy? 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
[OSL | CCIE_Voice] ESW and 3560
Hello Does anyone have spare HWIC-4ESW and 3560 I can buy? 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] HWIC-4ESW
Hi Hesham I was thinking of going that route as well. Do you know how different the configuration is when using a 3560/3750 instead of ESW. From what I read, the VLAN configuration will be slightly different, and I'll have to create a trunk as well between the branch router and branch switch. Thanks On Jul 25, 2013, at 1:20 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Sir, Just use 3550 24 Poe switch it cost $70 on ebay. That HWIC will cost u at least $200 or more. I know its better for practicing labs but its not cost effective. You can find it on ebay.com Thanks, Hesham On 25 July 2013 09:45, CCIE Voice Aspirant ccievoice2013.2...@gmail.com wrote: Hello list I am looking for 2 HWIC-4ESW cards for my lab, does anyone have spares I can buy? Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] HWIC-4ESW
Sounds good, thanks Hesham! On Jul 25, 2013, at 1:29 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Just get that model is good enough and cheap 3550-PWR-24 On 25 July 2013 11:28, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Yes sure just make DOT1Q on port 24 switchport mode trunk switchport trunk encapsulation dot1q in the Branch router like gig0/1 or whatever make a router on stick (Sub interfaces) Int gig0/0.302 encapsulation dot1q 302 ip address 142.102.65.254 255.255.255.0 no shut int gig0/0.402 encpasulation dot1q 402 ip address 142.202.65.254 255.255.255.0 no shut ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Channel Selection Order
Hi Guys, I am trying to understand the Channel selection order, i know that if TELCo is using bottom Up then i have to use opposite Direction i.e Top Down .My Telco is uses last channel for IN-bound calls and set Top down on my side for OUT-bound calls but still i can see that its taking bottom up for out bound calls,so did i miss any thing ? can any one share there knowledge and experience as to how i can configure it 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] Channel Selection Order
Thanks for ur reply,Usually the order configured in Cisco CallManager for outbound calls is the opposite of the Telco for inbound calls in order to increase the likelihood of available channels. On Wed, Jul 3, 2013 at 12:29 PM, LorenzLGRC lorenzl...@gmail.com wrote: Hello, as far as i know the B channel selection order is not bound to your telco configuration. You can set top-down or bottom-up with no issue at all. hth lorenz On Wed, Jul 3, 2013 at 10:58 AM, CISCO CCIE VOICE ccievoic...@gmail.comwrote: Hi Guys, I am trying to understand the Channel selection order, i know that if TELCo is using bottom Up then i have to use opposite Direction i.e Top Down .My Telco is uses last channel for IN-bound calls and set Top down on my side for OUT-bound calls but still i can see that its taking bottom up for out bound calls,so did i miss any thing ? can any one share there knowledge and experience as to how i can configure it Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 89, Issue 4
Hi Michal, As i can see that Clock source line command which is by default enable further can u give an example for both User Side and Network Side means can u past the commands that indicates both user and Network Side Thanks On Mon, Jul 1, 2013 at 7:00 PM, ccie_voice-requ...@onlinestudylist.comwrote: Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: How to configure Clocking for GW's (michael.se...@compucom.com) -- Message: 1 Date: Mon, 1 Jul 2013 15:09:31 + From: michael.se...@compucom.com To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] How to configure Clocking for GW's Message-ID: E88AFBC60579234C886F57D49AC243A937026EA9@SPW099EXM06.compucom.local Content-Type: text/plain; charset=us-ascii How to set clocking for PSTN Router and Branch Routers Example using MGCP gateway: ! ! Question: Take clocking for Layer 1 from Network side. --Means PSTN is Network Side Your PRI clocking of layer 2 should be user side. --Means Branch takes clock from PSTN ! ! PSTN ROUTER network-clock-participate wic 1 controller T1 0/1/0 clock source internal linecode B8ZS framing ESF pri-group timeslots 1-24 clock source internal Makes PSTN the Network Side description ** T1 PRI Voice Connection To: S0/1/0 BR1-RTR ** ! ! interface Serial0/1/0:23 description ** T1 VOICE CONNECTION TO S0/1/0 BR1-RTR ** no ip address encapsulation hdlc isdn switch-type primary-ni isdn protocol-emulate network Makes PSTN the Network Side isdn incoming-voice voice isdn negotiate-bchan resend-setup isdn outgoing display-ie isdn outgoing ie redirecting-number no cdp enable ! ! PSTN#show controller t1 0/1/0 T1 0/1/0 is up. Framing is ESF, Line Code is B8ZS, Clock Source is Internal. ! ! BRANCH ROUTERS USING T1 MGCP isdn switch-type primary-ni network-clock-participate wic 0 network-clock-select 1 T1 0/1/0 controller T1 0/0/0 pri-group timeslots 1-24 service mgcp clock source line Makes Router the user Side gets clock from PSTN linecode B8ZS framing ESF description ==VOICE PRI== ! ! interface Serial0/1/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn bind-l3 ccm-manager isdn send-alerting isdn sending-complete isdn outgoing display-ie isdn outgoing ie redirecting-number no cdp enable ! ! BR1-RTR#show controller t1 0/1/0 T1 0/1/0 is up. Framing is ESF, Line Code is B8ZS, Clock Source is Line. ! ! !Hope this helps clarify the clocking issues and configuration. ! ! Michael Sears, CCIE(V)#38404 Designing and Implementing Cisco Unified Communications on Unified Computing Systems -- ___ CCIE_Voice mailing list CCIE_Voice@onlinestudylist.com http://onlinestudylist.com/mailman/listinfo/ccie_voice End of CCIE_Voice Digest, Vol 89, Issue 4 * ___ 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] CCIE_Voice Digest, Vol 89, Issue 4
Thank You Michael, Question asked about L1 should me NETWORK SIDE and L2 USER SIDE in that case controller t1 0/0/0 clock source line -L1 Network Side (This is by default enable no need to add it) and for L2: USER SIDE do we need to add any additional commands under serial interface 0/0/0:23 ? Thanks On Mon, Jul 1, 2013 at 9:29 PM, Sears, Michael (msears) michael.se...@compucom.com wrote: NETWORK SIDE: ! PSTN ROUTER network-clock-participate wic 1 controller T1 0/1/0 clock source internal linecode B8ZS framing ESF pri-group timeslots 1-24 clock source internal Makes PSTN the Network Side description ** T1 PRI Voice Connection To: S0/1/0 BR1-RTR ** ! ! interface Serial0/1/0:23 description ** T1 VOICE CONNECTION TO S0/1/0 BR1-RTR ** no ip address encapsulation hdlc isdn switch-type primary-ni isdn protocol-emulate network Makes PSTN the Network Side isdn incoming-voice voice isdn negotiate-bchan resend-setup isdn outgoing display-ie isdn outgoing ie redirecting-number no cdp enable ! ! PSTN#show controller t1 0/1/0 T1 0/1/0 is up. Framing is ESF, Line Code is B8ZS, Clock Source is Internal. ! USER SIDE BRANCH ROUTERS USING T1 MGCP isdn switch-type primary-ni network-clock-participate wic 0 network-clock-select 1 T1 0/1/0 controller T1 0/0/0 pri-group timeslots 1-24 service mgcp clock source line Makes Router the user Side gets clock from PSTN linecode B8ZS framing ESF description ==VOICE PRI== ! ! interface Serial0/1/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn bind-l3 ccm-manager isdn send-alerting isdn sending-complete isdn outgoing display-ie isdn outgoing ie redirecting-number no cdp enable ! ! BR1-RTR#show controller t1 0/1/0 T1 0/1/0 is up. Framing is ESF, Line Code is B8ZS, Clock Source is Line. ! Michael Sears, CCIE(V)#38404 “Designing and Implementing Cisco Unified Communications on Unified Computing Systems” ** ** ___ 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] CCIE_Voice Digest, Vol 89, Issue 4
Yes, i am saying Layer 1 and Layer 2 of ISO Model On Mon, Jul 1, 2013 at 10:53 PM, Sears, Michael (msears) michael.se...@compucom.com wrote: When you say L1 and L2 are you talking about layer 1 and 2 of the OSI Model? ** ** Michael Sears, CCIE(V)#38404 “Designing and Implementing Cisco Unified Communications on Unified Computing Systems” ** ** *From:* CISCO CCIE VOICE [mailto:ccievoic...@gmail.com] *Sent:* Monday, July 01, 2013 12:58 PM *To:* Sears, Michael (msears) *Cc:* ccie_voice@onlinestudylist.com *Subject:* Re: CCIE_Voice Digest, Vol 89, Issue 4 ** ** Thank You Michael, Question asked about L1 should me NETWORK SIDE and L2 USER SIDE in that case ** ** controller t1 0/0/0 clock source line -L1 Network Side (This is by default enable no need to add it) ** ** and for L2: USER SIDE do we need to add any additional commands under serial interface 0/0/0:23 ? ** ** Thanks ** ** ** ** On Mon, Jul 1, 2013 at 9:29 PM, Sears, Michael (msears) michael.se...@compucom.com wrote: NETWORK SIDE: ! PSTN ROUTER network-clock-participate wic 1 controller T1 0/1/0 clock source internal linecode B8ZS framing ESF pri-group timeslots 1-24 clock source internal Makes PSTN the Network Side description ** T1 PRI Voice Connection To: S0/1/0 BR1-RTR ** ! ! interface Serial0/1/0:23 description ** T1 VOICE CONNECTION TO S0/1/0 BR1-RTR ** no ip address encapsulation hdlc isdn switch-type primary-ni isdn protocol-emulate network Makes PSTN the Network Side isdn incoming-voice voice isdn negotiate-bchan resend-setup isdn outgoing display-ie isdn outgoing ie redirecting-number no cdp enable ! ! PSTN#show controller t1 0/1/0 T1 0/1/0 is up. Framing is ESF, Line Code is B8ZS, Clock Source is Internal. ! USER SIDE BRANCH ROUTERS USING T1 MGCP isdn switch-type primary-ni network-clock-participate wic 0 network-clock-select 1 T1 0/1/0 controller T1 0/0/0 pri-group timeslots 1-24 service mgcp clock source line Makes Router the user Side gets clock from PSTN linecode B8ZS framing ESF description ==VOICE PRI== ! ! interface Serial0/1/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn bind-l3 ccm-manager isdn send-alerting isdn sending-complete isdn outgoing display-ie isdn outgoing ie redirecting-number no cdp enable ! ! BR1-RTR#show controller t1 0/1/0 T1 0/1/0 is up. Framing is ESF, Line Code is B8ZS, Clock Source is Line. ! Michael Sears, CCIE(V)#38404 “Designing and Implementing Cisco Unified Communications on Unified Computing Systems” ** ** ___ 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] CCIE_Voice Digest, Vol 88, Issue 131
Hi Todd, Please add the below commands and let me know the result... voice service voip allow-connections h323 to h323 h323 emptycapability h225 id-passthru h225 connect-passthru Thanks On Wed, Jun 26, 2013 at 5:38 AM, ccie_voice-requ...@onlinestudylist.comwrote: Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: Codec and CAC section (Karen Johnson) 2. Re: quto qos voip trust in WAN (Karen Johnson) 3. Five-Lab, Self-Study Challenge -- Lab #2 Gatekeeper (Todd Carswell) -- Message: 1 Date: Tue, 25 Jun 2013 09:12:49 -0700 (PDT) From: Karen Johnson karen.johnson...@yahoo.ca To: Amit Sharma aryan231...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Codec and CAC section Message-ID: 1372176769.52990.yahoomail...@web163901.mail.gq1.yahoo.com Content-Type: text/plain; charset=iso-8859-1 tks Amit, remembered seeing we do not need in DP. can someone confirm? From: Amit Sharma aryan231...@gmail.com To: Karen Johnson karen.johnson...@yahoo.ca Sent: Monday, June 24, 2013 10:49:38 PM Subject: Re: [OSL | CCIE_Voice] Codec and CAC section no... u have to add in both sides...! dont remove from DP...! On Mon, Jun 24, 2013 at 6:54 PM, Karen Johnson karen.johnson...@yahoo.ca wrote: hi amit, ? do you mean i should take Location out from DP? and just put in phones? ? tks K From: Amit Sharma aryan231...@gmail.com To: Karen Johnson karen.johnson...@yahoo.ca Sent: Monday, June 24, 2013 1:25:43 AM Subject: Re: [OSL | CCIE_Voice] Codec and CAC section u have to add 2 way sessions... and also add the location that should be apply to end phones...of hq and site-c. On Mon, Jun 24, 2013 at 9:07 AM, Karen Johnson karen.johnson...@yahoo.ca wrote: hi amit. ? why it need 8 instead of 4?? question ask for 4 g729 call only. ? tks From: Amit Sharma aryan231...@gmail.com To: Karen Johnson karen.johnson...@yahoo.ca Sent: Sunday, June 23, 2013 11:53:56 PM Subject: Re: [OSL | CCIE_Voice] Codec and CAC section dude you have to use this line: max session software 8.. this i think u missed and used 4 on behalf of 8... On Sun, Jun 23, 2013 at 9:26 PM, Karen Johnson karen.johnson...@yahoo.ca wrote: hi folks, ? can anyone share experience on what to check on?this section , I got 0 for few attempt. ? Here is what I did : ? UCM = ? - service parameter : no G722 and ILBC? - Enterprise parameter G711 intra, G729 inter - Region : HQ? SB?? SC,?? HQ-HQ : G711? , SB-SB? G711, SC-SC : g711?? (rest? relation : G729) ? and assign tp DP - Location : HQ? and SC? : mandatory , assign to DP - MRGL HQ -- MRG-- MTP from HQ??? ?? same for SC?? , assign to DP ? router HQ and SC = ? - dspfarm profile 3 mtp codec pass-through codec g729r8 rsvp maximum sessions software 4 (as they asked 4 session of g729) associate application SCCP - interface Serial0/0/0.1 point-to-point frame-relay interface-dlci 102 ip rsvp bandwidth 112 verification = - call hq to hq, sb sb : g711, inter site phone and GW : g729 - sh ip rsvp reservation : 40 k (ring) , and 24 k (connect) question: - did i miss something critical that cause the mark to be 0 ? ? ? ? ? ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.com/ Are you a CCNP or CCIE and looking for a job? Check out http://www.platinumplacement.com/ -- Thanks Regard'sAmit Sharma -- Thanks Regard'sAmit Sharma -- Thanks Regard'sAmit Sharma -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20130625/7976c600/attachment-0001.html -- Message: 2 Date: Tue, 25 Jun 2013 09:13:47 -0700 (PDT) From: Karen Johnson karen.johnson...@yahoo.ca To: Edgar Feliz ejzi...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] quto qos voip trust in WAN Message-ID: 1372176827.46135.yahoomail...@web163901.mail.gq1.yahoo.com Content-Type: text/plain; charset=iso-8859-1 hi edgar, ? I mean for WAN qos on Sb site in lab exam. do we need auto qos voip? or auto qos
Re: [OSL | CCIE_Voice] B-ACD
Thanks somphol i wll try that and let you know On Sun, Jun 23, 2013 at 11:56 AM, Somphol Boonjing somp...@gmail.comwrote: Remove all of the param under the service did the trick, Say you have this in your running config application service app-b-acd param number-of-hunt-grps 2 param aa-hunt1 param aa-hunt2 1222 param queue-len 15 param queue-manager-debugs 1 ! Then, application service app-b-acd no param number-of-hunt-grps 2 no param aa-hunt1 no param aa-hunt2 1222 no param queue-len 15 no param queue-manager-debugs 1 Once there is no param set for the service, it will be removed from the running-config. --- Detail trace below: --- Branch2#show run | begin application application service app-b-acd param queue-len 15 param aa-hunt1 param queue-manager-debugs 1 param aa-hunt2 1222 param number-of-hunt-grps 2 ! ! Branch2(config)#application Branch2(config-app)# service app-b-acd Branch2(config-app-param)#no param queue-len 15 Warning: parameter queue-len has not been registered under app-b-acd namespace Branch2(config-app-param)#no param aa-hunt1 Warning: parameter aa-hunt1 has not been registered under app-b-acd namespace Branch2(config-app-param)# Branch2(config-app-param)#do show run | begin application application service app-b-acd param queue-manager-debugs 1 param aa-hunt2 1222 param number-of-hunt-grps 2 ! ! Branch2(config-app-param)#no param queue-manager-debugs 1 Warning: parameter queue-manager-debugs has not been registered under app-b-acd namespace Branch2(config-app-param)#no param aa-hunt2 1222 Warning: parameter aa-hunt2 has not been registered under app-b-acd namespace Branch2(config-app-param)#no param number-of-hunt-grps 2 Warning: parameter number-of-hunt-grps has not been registered under app-b-acd namespace Branch2(config-app-param)#do show run | begin application associate application SCCP ! dspfarm profile 5 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 On Sun, Jun 23, 2013 at 12:20 AM, Bill Lake whl...@gmail.com wrote: Try doing all command not just these Sent from my iPhone On Jun 22, 2013, at 6:51 AM, CISCO CCIE VOICE ccievoic...@gmail.com wrote: Thanks Bill for your reply, I have done no service app-b-acd and no service app-b-acd-aa but showing all those commands in Running configuration thanks On Sat, Jun 22, 2013 at 1:48 PM, Bill Lake whl...@gmail.com wrote: If it is showing up in the running configuration, then you most likely see something like below, the best way to remove this is to no the commands Or to have done a Archive or copy of the config before you apply it. then restore that config as the startup and reboot. application * service app-b-acd * param number-of-hunt-grps 2 param aa-hunt2 param aa-hunt3 1222 param queue-len 15 param queue-manager-debugs 1 ! * service app-b-acd-aa * paramspace english index 1 paramspace english language en paramspace english location flash: param service-name app-b-acd param handoff-string app-b-acd-aa param aa-pilot 8005550123 param welcome-prompt _bacd_welcome.au param number-of-hunt-grps 2 param dial-by-extension-option 1 param second-greeting-time 60 param call-retry-timer 15 param max-time-call-retry 700 param max-time-vm-retry 2 param voice-mail 5003 ! dial-peer voice 222 voip service app-b-acd-aa destination-pattern 8005550123 session target ipv4:192.168.1.1 incoming called-number 8005550123 dtmf-relay h245-alphanumeric codec g711ulaw no vad On Sat, Jun 22, 2013 at 2:25 AM, Somphol Boonjing somp...@gmail.comwrote: That one is the embedded one so you actually can not remove it. However, you can simply ignore it and use one that is external script. So, if you have the external BACD script, you can use it instead of the embedded one. Branch2#show flash | inc bacd 107 30421bacd/app-b-acd-3.0.0.2.tcl 108 55599bacd/app-b-acd-aa-3.0.0.2.tcl application service *funnyqueue flash:/bacd/*app-b-acd-3.0.0.2.tcl -- you can you whatever name you like, in this case funnyqueue -- point the script to the script with correct path . (detail remove for brevity)... ! service* funnyaa flash:/bacd/app-b-acd-aa-3.0.0.2.tcl * -- you can you whatever name you like, in this case funnyaa -- point the script to the script with correct path . (detail remove for brevity). param service-name *funnyqueue* -- refer to your queue application name param handoff-string *funnyaa* . (detail remove for brevity). ! dial-peer voice 222 voip service *funnyaa* -- refer to your AA application name. . (detail remove for brevity
Re: [OSL | CCIE_Voice] B-ACD
application no service app-b-acd no service app-b-acd-aa On Sat, Jun 22, 2013 at 9:06 AM, Somphol Boonjing somp...@gmail.com wrote: Hi, Are you able to show part of the configuration that you have tried to remove from the running configuration? --Somphol On Sat, Jun 22, 2013 at 1:00 AM, CISCO CCIE VOICE ccievoic...@gmail.comwrote: Hi, I am trying to Remove B-ACD configuration but still showing in the running configuration i have restarted the router but no look any guess? thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] B-ACD
Hi Boonjing, thanks for ur reply,what if i have the requirement to use only embedded B-A-CD and if any reason i have to remove it then what i have to do ? On Sat, Jun 22, 2013 at 10:25 AM, Somphol Boonjing somp...@gmail.comwrote: That one is the embedded one so you actually can not remove it. However, you can simply ignore it and use one that is external script. So, if you have the external BACD script, you can use it instead of the embedded one. Branch2#show flash | inc bacd 107 30421bacd/app-b-acd-3.0.0.2.tcl 108 55599bacd/app-b-acd-aa-3.0.0.2.tcl application service *funnyqueue flash:/bacd/*app-b-acd-3.0.0.2.tcl -- you can you whatever name you like, in this case funnyqueue -- point the script to the script with correct path . (detail remove for brevity)... ! service* funnyaa flash:/bacd/app-b-acd-aa-3.0.0.2.tcl * -- you can you whatever name you like, in this case funnyaa -- point the script to the script with correct path . (detail remove for brevity). param service-name *funnyqueue* -- refer to your queue application name param handoff-string *funnyaa* . (detail remove for brevity). ! dial-peer voice 222 voip service *funnyaa* -- refer to your AA application name. . (detail remove for brevity)... ! To remove it from the running config, then you can, application no service *funnyqueue flash:/bacd/*app-b-acd-3.0.0.2.tcl no service* funnyaa flash:/bacd/app-b-acd-aa-3.0.0.2.tcl* Ref: http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html#wp1026305 Compared to using the embedded one below: application service app-b-acd -- you can't change the name of the embedded BACD Queue script . (detail remove for brevity)... ! service app-b-acd-aa -- you can't change the name of the embedded BACD AA script . (detail remove for brevity). param service-name app-b-acd -- refer to the embedded BACD Queue script param handoff-string app-b-acd-aa . (detail remove for brevity). ! dial-peer voice 222 voip service app-b-acd-aa -- refer to the name of the embedded BACD AA script . (detail remove for brevity)... ! Ref: Embedded Call-Queue and AA Tcl Scripts: Example http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html On 22/06/2013 4:18 PM, CISCO CCIE VOICE ccievoic...@gmail.com wrote: application no service app-b-acd no service app-b-acd-aa On Sat, Jun 22, 2013 at 9:06 AM, Somphol Boonjing somp...@gmail.comwrote: Hi, Are you able to show part of the configuration that you have tried to remove from the running configuration? --Somphol On Sat, Jun 22, 2013 at 1:00 AM, CISCO CCIE VOICE ccievoic...@gmail.com wrote: Hi, I am trying to Remove B-ACD configuration but still showing in the running configuration i have restarted the router but no look any guess? thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] B-ACD
Thanks Bill for your reply, I have done no service app-b-acd and no service app-b-acd-aa but showing all those commands in Running configuration thanks On Sat, Jun 22, 2013 at 1:48 PM, Bill Lake whl...@gmail.com wrote: If it is showing up in the running configuration, then you most likely see something like below, the best way to remove this is to no the commands Or to have done a Archive or copy of the config before you apply it. then restore that config as the startup and reboot. application * service app-b-acd * param number-of-hunt-grps 2 param aa-hunt2 param aa-hunt3 1222 param queue-len 15 param queue-manager-debugs 1 ! * service app-b-acd-aa * paramspace english index 1 paramspace english language en paramspace english location flash: param service-name app-b-acd param handoff-string app-b-acd-aa param aa-pilot 8005550123 param welcome-prompt _bacd_welcome.au param number-of-hunt-grps 2 param dial-by-extension-option 1 param second-greeting-time 60 param call-retry-timer 15 param max-time-call-retry 700 param max-time-vm-retry 2 param voice-mail 5003 ! dial-peer voice 222 voip service app-b-acd-aa destination-pattern 8005550123 session target ipv4:192.168.1.1 incoming called-number 8005550123 dtmf-relay h245-alphanumeric codec g711ulaw no vad On Sat, Jun 22, 2013 at 2:25 AM, Somphol Boonjing somp...@gmail.comwrote: That one is the embedded one so you actually can not remove it. However, you can simply ignore it and use one that is external script. So, if you have the external BACD script, you can use it instead of the embedded one. Branch2#show flash | inc bacd 107 30421bacd/app-b-acd-3.0.0.2.tcl 108 55599bacd/app-b-acd-aa-3.0.0.2.tcl application service *funnyqueue flash:/bacd/*app-b-acd-3.0.0.2.tcl -- you can you whatever name you like, in this case funnyqueue -- point the script to the script with correct path . (detail remove for brevity)... ! service* funnyaa flash:/bacd/app-b-acd-aa-3.0.0.2.tcl * -- you can you whatever name you like, in this case funnyaa -- point the script to the script with correct path . (detail remove for brevity). param service-name *funnyqueue* -- refer to your queue application name param handoff-string *funnyaa* . (detail remove for brevity). ! dial-peer voice 222 voip service *funnyaa* -- refer to your AA application name. . (detail remove for brevity)... ! To remove it from the running config, then you can, application no service *funnyqueue flash:/bacd/*app-b-acd-3.0.0.2.tcl no service* funnyaa flash:/bacd/app-b-acd-aa-3.0.0.2.tcl* Ref: http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html#wp1026305 Compared to using the embedded one below: application service app-b-acd -- you can't change the name of the embedded BACD Queue script . (detail remove for brevity)... ! service app-b-acd-aa -- you can't change the name of the embedded BACD AA script . (detail remove for brevity). param service-name app-b-acd -- refer to the embedded BACD Queue script param handoff-string app-b-acd-aa . (detail remove for brevity). ! dial-peer voice 222 voip service app-b-acd-aa -- refer to the name of the embedded BACD AA script . (detail remove for brevity)... ! Ref: Embedded Call-Queue and AA Tcl Scripts: Example http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html On 22/06/2013 4:18 PM, CISCO CCIE VOICE ccievoic...@gmail.com wrote: application no service app-b-acd no service app-b-acd-aa On Sat, Jun 22, 2013 at 9:06 AM, Somphol Boonjing somp...@gmail.comwrote: Hi, Are you able to show part of the configuration that you have tried to remove from the running configuration? --Somphol On Sat, Jun 22, 2013 at 1:00 AM, CISCO CCIE VOICE ccievoic...@gmail.com wrote: Hi, I am trying to Remove B-ACD configuration but still showing in the running configuration i have restarted the router but no look any guess? thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check
[OSL | CCIE_Voice] B-ACD issue
HI experts, I am trying to remove B-ACD cnofigs from the router but still show in running config,i have reloaded the router but still no luck any guess ? Thks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] B-ACD
Hi, I am trying to Remove B-ACD configuration but still showing in the running configuration i have restarted the router but no look any guess? 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] BACD Timer
Thanks Martin, i went through that doc but still its not clear to me the purpose of using it and how does it effect my B-ACD script On Tue, Jun 18, 2013 at 2:13 AM, Martin Sloan martinsloa...@gmail.comwrote: I got the info below from this guide - http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html#wp1026305 It has good examples you can copy/past/edit. I believe 'Call-Queue and AA Tcl Scripts in Flash Memory: Example' is the best one to use. http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html#wp1026305 Step 29 *param* *max-time-call-retry* *seconds * Example: Router(config-app-param)# param max-time-call-retry 700 (Optional) Sets the maximum amount of time for the call-retry timer. This is the maximum period of time for which a call can stay in a call queue and retry to connect with a hunt group before the call is sent to an alternate destination number. •*seconds*—Maximum period of time, in seconds. The range is from 30 to 3600. The default is 600. On Mon, Jun 17, 2013 at 5:11 PM, CISCO CCIE VOICE ccievoic...@gmail.comwrote: hi experts, I am trying to understand timer for *param* *max-time-call-retry can anyone share there knowledge about how does it effect the bacd script and the purpose of this field* * * *Thnks* * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] BACD Timer
hi experts, I am trying to understand timer for *param* *max-time-call-retry can anyone share there knowledge about how does it effect the bacd script and the purpose of this field* * * *Thnks* * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] UCCX Native codec g729
Hi, How can i verify that UCCX is using G729 codec native Thnajs ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] UCCX Native Codec G729
Hi, How can i verify that UCCX is using G729 codec native 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] UCCX Native codec g729
Thanks for your reply its fine that we have select G729 codec under System Parametes but my question was is there a way to test which shows UCCX is using g729 codec native On Sun, Jun 16, 2013 at 1:15 PM, OSL StudyList collaboration.c...@gmail.com wrote: Go to CCX admin and look under System and select System Parameters — Sent from Mailbox https://www.dropbox.com/mailbox for iPad On Sun, Jun 16, 2013 at 3:37 AM, CISCO CCIE VOICE ccievoic...@gmail.comwrote: Hi, How can i verify that UCCX is using G729 codec native Thnajs ___ 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] UCCX Native Codec G729
After i configure UCCX with codec g729 under service parameter,if i call to UCCX trigger number let say 5000 from HQ or Branch 1 or PSTN how do i know that this call has been using g729 uccx native codec not that from region On Sun, Jun 16, 2013 at 2:41 PM, Somphol Boonjing somp...@gmail.com wrote: Hi, I think the easiest way is to check UCCX Service Parameters under System menu. --Somphol. --Somphol On Sun, Jun 16, 2013 at 7:36 PM, CISCO CCIE VOICE ccievoic...@gmail.comwrote: Hi, How can i verify that UCCX is using G729 codec native Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] UCCX Native Codec G729
Thanks Bill,but if i Press question mark 2 time thn i can see its taking g729 codec that fine but ths result can be due to Region not from UCCX G729 Codec On Sun, Jun 16, 2013 at 3:37 PM, Bill Talley bill.tal...@aos5.com wrote: On the IP phone, press the question mark button twice. On the vgw, 'show call active voice compact'. *Bill Talley* UC Systems Consultant *Alexander Open Systems, Inc* 913.307.2330 (scheduling) | 913.744.3219 (direct) Web http://www.aos5.com/ | Request Support http://www.aos5.com/support | Facebookhttp://www.facebook.com/pages/Alexander-Open-Systems-AOS/109484829074064 | LinkedIn http://www.linkedin.com/company/aos Sent from an Apple iOS device with very tiny touchscreen input keys. Please excude my typtos. On Jun 16, 2013, at 8:14 AM, CISCO CCIE VOICE ccievoic...@gmail.com wrote: After i configure UCCX with codec g729 under service parameter,if i call to UCCX trigger number let say 5000 from HQ or Branch 1 or PSTN how do i know that this call has been using g729 uccx native codec not that from region On Sun, Jun 16, 2013 at 2:41 PM, Somphol Boonjing somp...@gmail.comwrote: Hi, I think the easiest way is to check UCCX Service Parameters under System menu. --Somphol. --Somphol On Sun, Jun 16, 2013 at 7:36 PM, CISCO CCIE VOICE ccievoic...@gmail.com wrote: Hi, How can i verify that UCCX is using G729 codec native Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com CONFIDENTIALITY NOTICE: This electronic mail transmission (including any accompanying attachments) is intended solely for its authorized recipient(s), and may contain confidential and/or legally privileged information. If you are not an intended recipient, or responsible for delivering some or all of this transmission to an intended recipient, be aware that any review, copying, printing, distribution, use or disclosure of the contents of this message is strictly prohibited. If you have received this electronic mail message in error, please delete it from your system without copying it, and contact sender immediately by Reply e-mail, or by calling 913-307-2300, so that our address records can be corrected. Although this e-mail and any attachments are believed to be free of any virus or other defect that might negatively affect any computer system into which it is received and opened, it is the responsibility of the recipient to ensure that it is virus free and no responsibility is accepted by the sender for any loss or damage arising in any way in the event that such a virus or defect exists. ___ 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] UCCX Native Codec G729
I have Branch 2 site which contains Transcoder moreover all my agents are in Branch-2 site i have region matrix between other region On Sun, Jun 16, 2013 at 4:54 PM, Bill Talley bill.tal...@aos5.com wrote: Do you have a transcoder setup and being used for the call? If so, check the SCCC connections. Otherwise you would get a busy signal if region said g729 and CCX was set for g711. Sent from an Apple iOS device with very tiny touchscreen input keys. Please excude my typtos. On Jun 16, 2013, at 9:41 AM, CISCO CCIE VOICE ccievoic...@gmail.com wrote: Thanks Bill,but if i Press question mark 2 time thn i can see its taking g729 codec that fine but ths result can be due to Region not from UCCX G729 Codec On Sun, Jun 16, 2013 at 3:37 PM, Bill Talley bill.tal...@aos5.com wrote: On the IP phone, press the question mark button twice. On the vgw, 'show call active voice compact'. *Bill Talley* UC Systems Consultant *Alexander Open Systems, Inc* 913.307.2330 (scheduling) | 913.744.3219 (direct) Web http://www.aos5.com/ | Request Supporthttp://www.aos5.com/support | Facebookhttp://www.facebook.com/pages/Alexander-Open-Systems-AOS/109484829074064 | LinkedIn http://www.linkedin.com/company/aos Sent from an Apple iOS device with very tiny touchscreen input keys. Please excude my typtos. On Jun 16, 2013, at 8:14 AM, CISCO CCIE VOICE ccievoic...@gmail.com wrote: After i configure UCCX with codec g729 under service parameter,if i call to UCCX trigger number let say 5000 from HQ or Branch 1 or PSTN how do i know that this call has been using g729 uccx native codec not that from region On Sun, Jun 16, 2013 at 2:41 PM, Somphol Boonjing somp...@gmail.comwrote: Hi, I think the easiest way is to check UCCX Service Parameters under System menu. --Somphol. --Somphol On Sun, Jun 16, 2013 at 7:36 PM, CISCO CCIE VOICE ccievoic...@gmail.com wrote: Hi, How can i verify that UCCX is using G729 codec native Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com CONFIDENTIALITY NOTICE: This electronic mail transmission (including any accompanying attachments) is intended solely for its authorized recipient(s), and may contain confidential and/or legally privileged information. If you are not an intended recipient, or responsible for delivering some or all of this transmission to an intended recipient, be aware that any review, copying, printing, distribution, use or disclosure of the contents of this message is strictly prohibited. If you have received this electronic mail message in error, please delete it from your system without copying it, and contact sender immediately by Reply e-mail, or by calling 913-307-2300, so that our address records can be corrected. Although this e-mail and any attachments are believed to be free of any virus or other defect that might negatively affect any computer system into which it is received and opened, it is the responsibility of the recipient to ensure that it is virus free and no responsibility is accepted by the sender for any loss or damage arising in any way in the event that such a virus or defect exists. CONFIDENTIALITY NOTICE: This electronic mail transmission (including any accompanying attachments) is intended solely for its authorized recipient(s), and may contain confidential and/or legally privileged information. If you are not an intended recipient, or responsible for delivering some or all of this transmission to an intended recipient, be aware that any review, copying, printing, distribution, use or disclosure of the contents of this message is strictly prohibited. If you have received this electronic mail message in error, please delete it from your system without copying it, and contact sender immediately by Reply e-mail, or by calling 913-307-2300, so that our address records can be corrected. Although this e-mail and any attachments are believed to be free of any virus or other defect that might negatively affect any computer system into which it is received and opened, it is the responsibility of the recipient to ensure that it is virus free and no responsibility is accepted by the sender for any loss or damage arising in any way in the event that such a virus or defect exists. ___ 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] UCCX Native Codec G729
Thanks bill got your point On Sun, Jun 16, 2013 at 5:13 PM, Bill Talley bill.tal...@aos5.com wrote: Any transcoder for ccx would be at HQ unless the ccx server is not in HQ. looks like a few people have forwarded viable suggestions so you should be able to determine what you're asking fairly quickly. Good luck. Sent from an Apple iOS device with very tiny touchscreen input keys. Please excude my typtos. On Jun 16, 2013, at 10:06 AM, CISCO CCIE VOICE ccievoic...@gmail.com wrote: I have Branch 2 site which contains Transcoder moreover all my agents are in Branch-2 site i have region matrix between other region On Sun, Jun 16, 2013 at 4:54 PM, Bill Talley bill.tal...@aos5.com wrote: Do you have a transcoder setup and being used for the call? If so, check the SCCC connections. Otherwise you would get a busy signal if region said g729 and CCX was set for g711. Sent from an Apple iOS device with very tiny touchscreen input keys. Please excude my typtos. On Jun 16, 2013, at 9:41 AM, CISCO CCIE VOICE ccievoic...@gmail.com wrote: Thanks Bill,but if i Press question mark 2 time thn i can see its taking g729 codec that fine but ths result can be due to Region not from UCCX G729 Codec On Sun, Jun 16, 2013 at 3:37 PM, Bill Talley bill.tal...@aos5.comwrote: On the IP phone, press the question mark button twice. On the vgw, 'show call active voice compact'. *Bill Talley* UC Systems Consultant *Alexander Open Systems, Inc* 913.307.2330 (scheduling) | 913.744.3219 (direct) Web http://www.aos5.com/ | Request Supporthttp://www.aos5.com/support | Facebookhttp://www.facebook.com/pages/Alexander-Open-Systems-AOS/109484829074064 | LinkedIn http://www.linkedin.com/company/aos Sent from an Apple iOS device with very tiny touchscreen input keys. Please excude my typtos. On Jun 16, 2013, at 8:14 AM, CISCO CCIE VOICE ccievoic...@gmail.com wrote: After i configure UCCX with codec g729 under service parameter,if i call to UCCX trigger number let say 5000 from HQ or Branch 1 or PSTN how do i know that this call has been using g729 uccx native codec not that from region On Sun, Jun 16, 2013 at 2:41 PM, Somphol Boonjing somp...@gmail.comwrote: Hi, I think the easiest way is to check UCCX Service Parameters under System menu. --Somphol. --Somphol On Sun, Jun 16, 2013 at 7:36 PM, CISCO CCIE VOICE ccievoic...@gmail.com wrote: Hi, How can i verify that UCCX is using G729 codec native Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com CONFIDENTIALITY NOTICE: This electronic mail transmission (including any accompanying attachments) is intended solely for its authorized recipient(s), and may contain confidential and/or legally privileged information. If you are not an intended recipient, or responsible for delivering some or all of this transmission to an intended recipient, be aware that any review, copying, printing, distribution, use or disclosure of the contents of this message is strictly prohibited. If you have received this electronic mail message in error, please delete it from your system without copying it, and contact sender immediately by Reply e-mail, or by calling 913-307-2300, so that our address records can be corrected. Although this e-mail and any attachments are believed to be free of any virus or other defect that might negatively affect any computer system into which it is received and opened, it is the responsibility of the recipient to ensure that it is virus free and no responsibility is accepted by the sender for any loss or damage arising in any way in the event that such a virus or defect exists. CONFIDENTIALITY NOTICE: This electronic mail transmission (including any accompanying attachments) is intended solely for its authorized recipient(s), and may contain confidential and/or legally privileged information. If you are not an intended recipient, or responsible for delivering some or all of this transmission to an intended recipient, be aware that any review, copying, printing, distribution, use or disclosure of the contents of this message is strictly prohibited. If you have received this electronic mail message in error, please delete it from your system without copying it, and contact sender immediately by Reply e-mail, or by calling 913-307-2300, so that our address records can be corrected. Although this e-mail and any attachments are believed to be free of any virus or other defect that might negatively affect any computer
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 87, Issue 56
Hi experts, i am also searching for that solution thanks i will also try that and let u know, which command i need to run on SB gateway to get the output,do i need call start fast under voice service voip in SB gateway ? Thanks On Tue, May 21, 2013 at 11:41 AM, ccie_voice-requ...@onlinestudylist.comwrote: Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: CCIE_Voice Digest, Vol 87, Issue 55 (Mohammed Ameenullah) -- Message: 1 Date: Tue, 21 May 2013 11:41:39 +0300 From: Mohammed Ameenullah ameen...@gmail.com To: ccie_voice@onlinestudylist.com Cc: jainpiyush2...@ymail.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 87, Issue 55 Message-ID: caau-xfjcw1d08ac3hydp2npcgy_jzq_au8kcsulzeeh549z...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi Piyush, i have tried with with G711ulaw on SB gateway its working fine for me with redundant to HQ call routing here what i have done I have created MTP and Xcode on Site B router sccp ccm group 1 ass ccm 1 prio 1 ass ccm 2 prio 2 ass pro 1 reg SB-XCODE ass pro 2 reg SB-MTP dspfarm pro 1 trans max sess 4 ass app sccp no shut dspfram pro 2 mtp codec g711ulaw max sess soft 8 ass app sccp no shut and on CUCM u have to create MRG n MRGL and assign ths MRGL to SB Gateway and check MTP required in CUCM Gateway page you can try ths configuration and let me know ur feedback ... On Tue, May 21, 2013 at 10:08 AM, ccie_voice-requ...@onlinestudylist.com wrote: 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. (no subject) (ie ravindra) 2. (MGCP Teardown) (ie ravindra) 3. Re: (no subject) (Shabeeb Mohammed) 4. h323 Fast start configuration (Piyush Jain) -- Message: 1 Date: Tue, 21 May 2013 04:26:42 +0530 From: ie ravindra ieravin...@gmail.com To: CCIE Study ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] (no subject) Message-ID: cabadenywv08supms0meguj29ihtrbt+enrvpouse56vcyxu...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Dear All, Whats is the real meaning of MGCP tear down. Is it means dropping a call or , What ? thanks for your valuable input. Ravi, -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20130521/91fde8a8/attachment-0001.html -- Message: 2 Date: Tue, 21 May 2013 04:41:48 +0530 From: ie ravindra ieravin...@gmail.com To: CCIE Study ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] (MGCP Teardown) Message-ID: CABADEnz40hKrB=bWpHQA8eNomj3v6nU= 57ufmvq+hnvmmm-...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Dear All, Whats is the real meaning of MGCP tear down. Is it means dropping a call or , What ? thanks for your valuable input. Ravi, -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20130521/c6e48219/attachment-0001.html -- Message: 3 Date: Tue, 21 May 2013 10:36:49 +0530 From: Shabeeb Mohammed shabeebc...@gmail.com To: ie ravindra ieravin...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] (no subject) Message-ID: caojzbky0gwx7s_2udiduvw6euuqsfxfuwzm6638mtg1a+pn...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hey ravi, I believe it means that mgcp packets was disrupted in between transmission resulting in packet loss etc. This error implies that there is a bug in the ios. Try upgrading the ios and check Regards Shabeeb On 21 May 2013 04:28, ie ravindra ieravin...@gmail.com wrote: Dear All, Whats is the real meaning of MGCP tear
Re: [OSL | CCIE_Voice] NO Extension in CME-SRST
Hi, Command button is showing in the configuration but the extension is not appearing on the phone thnks On Fri, May 17, 2013 at 1:31 AM, Vignesh Sethuraman sethuvign...@gmail.comwrote: Please check if the command button x:x is available or not. It might have got removed. On Thu, May 16, 2013 at 10:21 PM, CISCO CCIE VOICE ccievoic...@gmail.comwrote: HI experts, When the Phones are in CME SRST its does not shows Extension on Phone display this happens when i change the name +86223033001 to SiteC Phone 1 under Ephone-dn's Thnks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] NO Extension in CME-SRST
HI experts, When the Phones are in CME SRST its does not shows Extension on Phone display this happens when i change the name +86223033001 to SiteC Phone 1 under Ephone-dn's Thnks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Sending Calling Name to PSTN
Hi Experts, How can i send Calling name to PSTN over MGCP and H323 Gateway? 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
[OSL | CCIE_Voice] Unable to play welcome Prompt
Hi Experts, When i dial UCCX Trigger i am not able to play custom welcome prompt rather its playing MOH. 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
[OSL | CCIE_Voice] IPCC Problem
Hi Experts, I am getting an error i am sorry we r currently facing system problm and r unable t process ur call during CSQ 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] IPCC Problem
Hi, whn i call frm PSTN to 2022400 it play the Thank you for calling then Thank You for waiting, we wll connect you with our Agent HQ1 then i am getting MOH audio after that call will goes to Site B Phone 1 and play the prompt as Thank You for waiting, we wll connect you with our Agent SiteB1 and then i am able hear MOH audio .After that i getting i am sorry we r currently facing system problm and r unable t process ur call On Tue, Apr 30, 2013 at 9:26 AM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Thats definitely a wrong script was made however , If you want to update the script make the following steps 1-Edit the script on the UCCX Editor Application 2-Save As anything (the old name same as listed on UCCX Admin) 3-Validate the script on UCCX Editor and make sure its successfully validated 4-Then go to UCCX Admin Script Management - Upload the Script -- Overwrite the Script -- Then Refresh The Script 5-Refresh the Application Script Then you should be good to go Thanks, Hesham On 29 April 2013 23:03, CISCO CCIE VOICE ccievoic...@gmail.com wrote: Hi Experts, I am getting an error i am sorry we r currently facing system problm and r unable t process ur call during CSQ Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] IPCC Problem
I am facing the same problem when is use default icd.aef script ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] SIP Delay Vs Early Offer
Thanks Robert On Sun, Apr 28, 2013 at 5:11 PM, Robert Thomas tho...@gmail.com wrote: The description is not entirely correct. Delayed Offer: The UAC SDP is relayed on the ACK Message INVITE - 100 Trying - 200OK - With SDP ACK - With SDP Early Offer INVITE - With SDP 100 Trying - 200OK - With SDP ACK - The SDP is moved from the ACK to the INVITE, This is why is Early Offer. The 200OK should always contain SDP from the UAS. Only Exception might be when the SDP is sent on a provisional 1XX Message, like a 180 Ringing or 183 Session Progress. In that case is refered to as Early Media instead of early offer. So saying on delayed offer the SDP is on the 200OK, is incorrect since, both scenarios have SDP on the 200OK, The difference is the variation between the INVITE and the ACK. This link has further description. http://www.iplogos.fr/English-Resources/Focus/sip-early-media-early-offer-en.html On Sun, Apr 28, 2013 at 2:51 AM, CISCO CCIE VOICE ccievoic...@gmail.comwrote: Thank you suresh i will check ths on my pod... On Sun, Apr 28, 2013 at 11:26 AM, Suresh Bhandari bring...@gmail.comwrote: Content Length 0 means, there will be an OK followed with SDP So basically both are same. On Sun, Apr 28, 2013 at 1:56 PM, CISCO CCIE VOICE ccievoic...@gmail.com wrote: Hi Suresh, Thanks for you reply, If SIP 200 OK message has media then its DELAYED OFFER and SIP INVITE Message with content-length 0 also means the DELAYED OFFER so which one i have to consider ? thanks On Sun, Apr 28, 2013 at 10:59 AM, Suresh Bhandari bring...@gmail.comwrote: If your SIP Invite message is containing media information, it is an early offer. If the response OK message has media, then it is delayed offer. I prefer checking Content-Length field to find out if its an Early or Delayed offer. Value of 0 mean Delayed offer, non-zero value means it is including SDP message and so its an early offer. HTH On Sun, Apr 28, 2013 at 1:15 PM, CISCO CCIE VOICE ccievoic...@gmail.com wrote: HI Experts, Which SIP Message contains Delay Offer and Early Offer message ? i i am bit confuse as some documents says SIP INVITE message contains both Delay and Early Offer and some documents say its SIP 200 OK message 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 -- Suresh Bhandari -- 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 -- Robert Thomas Zamora 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
[OSL | CCIE_Voice] MGCP Auto vs Manual configuration
Hi Experts Which method to be use in CCIE Voice lab exam in order to configure MGCP Auto or Manually. 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] MGCP Auto vs Manual configuration
HI Thanks, but if suppose i want to use 6 channel from T1 PRI on HQ Router and 12 channel of E1 PRI from Branch-2 router,so how can i busy out channels under service parameter ? and how does CUCM knows about which router interface to busy out as i have two site with different fractional PRI Any reference or document on ths will be g8rt help... thanks On Sun, Apr 28, 2013 at 9:46 AM, Mohamed Gazzaz mgaz...@hotmail.com wrote: Pick up whatever method you are comfortable with and practice, practice. practice .. Try both methods and see which one is faster for you. -- Date: Sun, 28 Apr 2013 09:10:30 +0300 From: ccievoic...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] MGCP Auto vs Manual configuration Hi Experts Which method to be use in CCIE Voice lab exam in order to configure MGCP Auto or Manually. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] SIP Delay Vs Early Offer
HI Experts, Which SIP Message contains Delay Offer and Early Offer message ? i i am bit confuse as some documents says SIP INVITE message contains both Delay and Early Offer and some documents say its SIP 200 OK message 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] SIP Delay Vs Early Offer
Hi Suresh, Thanks for you reply, If SIP 200 OK message has media then its DELAYED OFFER and SIP INVITE Message with content-length 0 also means the DELAYED OFFER so which one i have to consider ? thanks On Sun, Apr 28, 2013 at 10:59 AM, Suresh Bhandari bring...@gmail.comwrote: If your SIP Invite message is containing media information, it is an early offer. If the response OK message has media, then it is delayed offer. I prefer checking Content-Length field to find out if its an Early or Delayed offer. Value of 0 mean Delayed offer, non-zero value means it is including SDP message and so its an early offer. HTH On Sun, Apr 28, 2013 at 1:15 PM, CISCO CCIE VOICE ccievoic...@gmail.comwrote: HI Experts, Which SIP Message contains Delay Offer and Early Offer message ? i i am bit confuse as some documents says SIP INVITE message contains both Delay and Early Offer and some documents say its SIP 200 OK message 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 -- 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] MGCP Auto vs Manual configuration
HI Thanks Again, So i can use Fractional T1/E1 PRI with Automatic MGCP Configuration? On Sun, Apr 28, 2013 at 10:56 AM, Suresh Bhandari bring...@gmail.comwrote: Regarding your question about busyout channels in CUCM, you will use the MGCP gateway address as appeared in your CUCM gateway page and busyout channels using 1. You can find a good example by clicking the parameter. For your reference, for say your SA site, the SP Change B-Channel Maintenance Status 1 will be S0/SU0/DS1-0@*SA-RTR* = 0011 to busyout channels 7-23. Notice the boldface text that represents your gateway. Further, you have to check Enable status Poll in your gateway page. Hope this helps. On Sun, Apr 28, 2013 at 12:45 PM, CISCO CCIE VOICE ccievoic...@gmail.comwrote: HI Thanks, but if suppose i want to use 6 channel from T1 PRI on HQ Router and 12 channel of E1 PRI from Branch-2 router,so how can i busy out channels under service parameter ? and how does CUCM knows about which router interface to busy out as i have two site with different fractional PRI Any reference or document on ths will be g8rt help... thanks On Sun, Apr 28, 2013 at 9:46 AM, Mohamed Gazzaz mgaz...@hotmail.comwrote: Pick up whatever method you are comfortable with and practice, practice. practice .. Try both methods and see which one is faster for you. -- Date: Sun, 28 Apr 2013 09:10:30 +0300 From: ccievoic...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] MGCP Auto vs Manual configuration Hi Experts Which method to be use in CCIE Voice lab exam in order to configure MGCP Auto or Manually. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- 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] SIP Delay Vs Early Offer
Thank you suresh i will check ths on my pod... On Sun, Apr 28, 2013 at 11:26 AM, Suresh Bhandari bring...@gmail.comwrote: Content Length 0 means, there will be an OK followed with SDP So basically both are same. On Sun, Apr 28, 2013 at 1:56 PM, CISCO CCIE VOICE ccievoic...@gmail.comwrote: Hi Suresh, Thanks for you reply, If SIP 200 OK message has media then its DELAYED OFFER and SIP INVITE Message with content-length 0 also means the DELAYED OFFER so which one i have to consider ? thanks On Sun, Apr 28, 2013 at 10:59 AM, Suresh Bhandari bring...@gmail.comwrote: If your SIP Invite message is containing media information, it is an early offer. If the response OK message has media, then it is delayed offer. I prefer checking Content-Length field to find out if its an Early or Delayed offer. Value of 0 mean Delayed offer, non-zero value means it is including SDP message and so its an early offer. HTH On Sun, Apr 28, 2013 at 1:15 PM, CISCO CCIE VOICE ccievoic...@gmail.com wrote: HI Experts, Which SIP Message contains Delay Offer and Early Offer message ? i i am bit confuse as some documents says SIP INVITE message contains both Delay and Early Offer and some documents say its SIP 200 OK message 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 -- Suresh Bhandari -- 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
[OSL | CCIE_Voice] ISDN No alerting Message
HI experts, When Branch-2 MGCP router is in CME SRST mode ,i am unable to see Alerting message on inbound call from PSTN,moreover when i call from PSTN my call is directly connected and there is no ring back tone ,Can any one help to fix the issue. ISDN Se0/0/0:15 Q931: RX - SETUP pd = 8 callref = 0x008D Bearer Capability i = 0x8090A3 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA18381 Preferred, Channel 1 Progress Ind i = 0x8183 - Origination address is non-ISDN Display i = 'Emergency 911/999' Calling Party Number i = 0x0180, '911' Plan:ISDN, Type:Unknown Called Party Number i = 0x81, '24044001' Plan:ISDN, Type:Unknown ISDN Se0/0/0:15 Q931: Received SETUP callref = 0x808D callID = 0x000D switch = primary-net5 interface = User ISDN Se0/0/0:15 Q931: TX - CALL_PROC pd = 8 callref = 0x808D Channel ID i = 0xA98381 Exclusive, Channel 1 ISDN Se0/0/0:15 Q931: TX - CONNECT pd = 8 callref = 0x808D ISDN Se0/0/0:15 Q931: RX - CONNECT_ACK pd = 8 callref = 0x008D Branch2(config-if)# %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now connected to 911 N/A %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now connected to 911 N/A ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] H323 and SLRG
Hi Guys, Can i group H323 GW and SLRG in one Route List ? it that the best practice ?can any one provide me with the reference for best practice to which protocol in can group together like h323, mgcp,sip trunk,GK Trunk ect in to Route List. 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
[OSL | CCIE_Voice] E1 Framing
Hi Guys, Regarding E1 Framing if the question states the below FOR E1 use the following: switch type:net5 framing :CRC linecode :HDB3 then do i need to use the NO-CRC or CRC ? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] IE display vs display IE
Hi guys, what is the difference between IE display and display IE command interface Serial0/1/0:15 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn incoming-voice voice isdn bind-l3 ccm-manager isdn outgoing *display-ie VS ie dislay* isdn outgoing ie redirecting-number no cdp enable ! 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] IE display vs display IE
Thanks All On Mon, Apr 1, 2013 at 2:48 PM, ie ravindra ieravin...@gmail.com wrote: I have seen this also. There are three commands. isdn outgoing display-ie / isdn outgoing ie display both commands are same as I heard. but there is a another command called facility ie. don't know what the use is. I have read in many document to use facility ie if display ie or ie display not worked. On Mon, Apr 1, 2013 at 5:06 PM, Suresh Bhandari bring...@gmail.comwrote: The help output displays the same thing: *isdn outgoing ?* display-ie DISPLAY IE in outgoing ISDN messages is allowed and *isdn outgoing ie ? * ... displayDISPLAY IE in outgoing ISDN messages is allowed ... so this may be otherwise option for the same display-ie. Any other views? On Mon, Apr 1, 2013 at 5:06 PM, CISCO CCIE VOICE ccievoic...@gmail.comwrote: Hi guys, what is the difference between IE display and display IE command interface Serial0/1/0:15 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn incoming-voice voice isdn bind-l3 ccm-manager isdn outgoing *display-ie VS ie dislay* isdn outgoing ie redirecting-number no cdp enable ! 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 -- 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
[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
[OSL | CCIE_Voice] CFUR
Hi experts, i have been trying to test the Call forward unregister whn the call arrive on destination phone its show on the display as FORWARD,FOR and BY on the phone screen so is there a way to display FROM on the destination phone screen 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
[OSL | CCIE_Voice] CUCM DHCP issue
Hi experts, i am facing a problem with CUCM DHCP ,my phones are taking an ip address as first IP from starting range and so on actually it has to take last ip first and so on eg: CUCM DHCP range :177.1.11.10 -177.1.11.30 MY phones getting an ip as 177.1.11.10 thanks in advance ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Directory folder on Router
Hi Experts, can any one help me I want to create directory folder on router without formatting flash when i use mkdir command its saying that invalid input thnks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Directory folder on Router
THANKS CORY AND BELL REALLY APPRECIATED On Tue, Mar 12, 2013 at 6:06 PM, Cory Gray corygray22...@hotmail.comwrote: I tried this for hours and there is no way (that I could find). You must format flash to get the command http://www.cisco.com/en/US/docs/routers/access/1800/1841/software/configuration/guide/b_cflash.html#wp23144 ** ** ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *CISCO CCIE VOICE *Sent:* Tuesday, March 12, 2013 10:51 AM *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] Directory folder on Router ** ** Hi Experts, ** ** can any one help me I want to create directory folder on router without formatting flash when i use mkdir command its saying that invalid input * *** ** ** thnks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] H323 Trunk to PSTN
HI experts, If the question ask to configure H323 Trunk to backbone router then in that do i need to configure it as H323 Gateway or H323 Non -GK controlled trunk? Thnks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] H323 Trunk Debugs in RTMT
Hi Experts, Can anyone explain me how to collect h323 trunk debugs from RTMT ... 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
[OSL | CCIE_Voice] SIP and H323 Trunk to PSTN
HI Guys, Can any one Share H323 and SIP Trunk configuration that need to be done on PSTN Router in order for it to work properly ... 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] CCIE_Voice Digest, Vol 85, Issue 11
Phone NTP reference is used for phones that uses SIP protocol to get there time ... On Tue, Mar 5, 2013 at 1:15 PM, ccie_voice-requ...@onlinestudylist.comwrote: Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Phone NTP Reference Vs NTP Reference (CCIEing) 2. Re: Phone NTP Reference Vs NTP Reference (Jason Lee) 3. Re: Phone NTP Reference Vs NTP Reference (Cory Gray) 4. Re: Phone NTP Reference Vs NTP Reference (CCIEing) 5. AAR HELP (CISCO CCIE VOICE) 6. Re: AAR HELP (Jamie Parr (jamparr)) 7. Re: AAR HELP (CISCO CCIE VOICE) -- Message: 1 Date: Tue, 5 Mar 2013 00:15:37 +0300 From: CCIEing aboaz...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Phone NTP Reference Vs NTP Reference Message-ID: CAEHpKk+Lkk6bve2AygUMTSEH=- fjd++6ubszgxyhqepodbb...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hello All, The following question cross my mind while doing the NTP configuration stuff.. What is the difference between the Phone NTP reference configuration in the CCM Web administration page and The NTP reference on the OS Administration page?? does the 1st one for the endpoints where the 2nd one is for the CUCM itself? Thanks -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20130305/981b33f7/attachment-0001.html -- Message: 2 Date: Mon, 4 Mar 2013 16:48:44 -0500 From: Jason Lee jas7...@gmail.com To: CCIEing aboaz...@gmail.com Cc: ccie_voice ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Phone NTP Reference Vs NTP Reference Message-ID: capf5dnpx9u2jpau501_xkedvcnemmtysrouh6+obvyfg0lg...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 The Phone NTP Reference is used for SIP endpoints. SIP endpoints store a NTP server address internally and they use the Phone NTP Reference parameter to obtain that information. This parameter is not required for SCCP endpoints. The second is for the CUCM server. You were pretty much spot on with your guess. HTH, Jason On Mon, Mar 4, 2013 at 4:15 PM, CCIEing aboaz...@gmail.com wrote: Hello All, The following question cross my mind while doing the NTP configuration stuff.. What is the difference between the Phone NTP reference configuration in the CCM Web administration page and The NTP reference on the OS Administration page?? does the 1st one for the endpoints where the 2nd one is for the CUCM itself? Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20130304/a63b29ef/attachment-0001.html -- Message: 3 Date: Mon, 4 Mar 2013 16:48:55 -0500 From: Cory Gray corygray22...@hotmail.com To: CCIEing aboaz...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Phone NTP Reference Vs NTP Reference Message-ID: bay404-eas56c8a84c06f6f0e59b0d5cfb...@phx.gbl Content-Type: text/plain; charset=us-ascii Phone ntp reference is for SIP phones only Sent from my iPhone On Mar 4, 2013, at 4:42 PM, CCIEing aboaz...@gmail.com wrote: Hello All, The following question cross my mind while doing the NTP configuration stuff.. What is the difference between the Phone NTP reference configuration in the CCM Web administration page and The NTP reference on the OS Administration page?? does the 1st one for the endpoints where the 2nd one is for the CUCM itself? Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- Message: 4 Date: Tue, 5 Mar 2013 01:05:22 +0300 From: CCIEing aboaz...@gmail.com To: Cory Gray corygray22...@hotmail.com, Suresh Bhandari bring...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Phone NTP Reference Vs NTP
[OSL | CCIE_Voice] VM in SRST
HI All When the Branch-1 is in SRST Proper Mode (call-manager-fallback) When i am trying t press voice mail button its playing system greeting not the user greeting ,its there any thing i need do on CUCM in order to play user greeting when the phones are in SRST.. 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
[OSL | CCIE_Voice] AAR HELP
Hi experts, Can any one help me to understand the below Question Call from HQ to Branch-2 should invoke AAR base on RSVP bandwidth,If HQPH1 calls BR2PH1 then it should display E.164 number and Network congestion Re-routing Message on Phone So ,my doubt is that do i need to use AAR and location on VM Pilot Number,VM Ports ,CUE-PORTS and CTI-Route Point etc 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] AAR HELP
Hi Jamie, Thanks for your reply,Actually the question does not provide much information its just say that Call From HQ to BRABCH-2 so is it safe to use on AAR Group/CSS as well as location on Voice Mail configuration let me try these Thanks On Tue, Mar 5, 2013 at 12:13 PM, Jamie Parr (jamparr) jamp...@cisco.comwrote: I configure AAR on all of these. Applying AAR to the device pool will mean only giving all the relevant extensions the AAR group, not much extra config as you have to visit all those pages anyway. Better safe than sorry ** ** *Jamie Parr* Engineer - IT jamp...@cisco.com Phone: *+44 20 8824 2641* Mobile: *+44 7590622049* ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *CISCO CCIE VOICE *Sent:* 05 March 2013 08:13 *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] AAR HELP ** ** Hi experts, Can any one help me to understand the below Question Call from HQ to Branch-2 should invoke AAR base on RSVP bandwidth,If HQPH1 calls BR2PH1 then it should display E.164 number and Network congestion Re-routing Message on Phone So ,my doubt is that do i need to use AAR and location on VM Pilot Number,VM Ports ,CUE-PORTS and CTI-Route Point etc 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
[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] Using BAT in LAB
Thanks Mark for quick reply... On Tue, Feb 26, 2013 at 11:50 PM, Mark Thrash (marthras) marth...@cisco.com wrote: Oh yea __ Mark Thrash 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.com Date: Tuesday, February 26, 2013 2:44 PM To: ccie_voice@onlinestudylist.com 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
[OSL | CCIE_Voice] MWI in SRST
HI, Does MWI works on SRST Proper mode (call-manager-fallback).? 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
[OSL | CCIE_Voice] Dail Plan Consideration in SRST Mode
Hi Can any one help me with Dial Plan consideration when calling from HQ Site to Branch 1 Site,following what i have configure.But the problem is that on B1PH1 screen its showing as below *From +14082021001 Forward by:2001* *HQ SITE:* *Extension Range*:1XXX *Partition:*Branch_1_SRST_PT *CSS :*Branch_1_SRST_CSS--contains-Branch_1_SRST_PT *B1PH1*:i have assign CFUR as 2001 with CSS:Branch_1_SRST_CSS *Route Pattern:* 2XXX--Branch_1_SRST_PT *Route List :* Standard Local Route Group *Prefix :*91972303 *BRANCH 1 SITE:* *Extension Range :*2XXX dial-peer voice 10 pots destination-pattern 1... perfix 14082021 port 0/0/0:23 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] Dail Plan Consideration in SRST Mode
Thanks i will configure as mention and let you know... On Sun, Feb 24, 2013 at 4:18 PM, Cory Gray corygray22...@hotmail.comwrote: You probably have redirecting number outbound checked on your site a gateway. Uncheck it, reset your gateway and let us know Sent from my iPhone On Feb 24, 2013, at 6:45 AM, CISCO CCIE VOICE ccievoic...@gmail.com wrote: Hi Can any one help me with Dial Plan consideration when calling from HQ Site to Branch 1 Site,following what i have configure.But the problem is that on B1PH1 screen its showing as below *From +14082021001 Forward by:2001* *HQ SITE:* *Extension Range*:1XXX *Partition:*Branch_1_SRST_PT *CSS :* Branch_1_SRST_CSS--contains-Branch_1_SRST_PT *B1PH1*:i have assign CFUR as 2001 with CSS:Branch_1_SRST_CSS *Route Pattern:* 2XXX--Branch_1_SRST_PT *Route List :* Standard Local Route Group *Prefix :*91972303 *BRANCH 1 SITE:* *Extension Range :*2XXX dial-peer voice 10 pots destination-pattern 1... perfix 14082021 port 0/0/0:23 thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] SRST DIAL PLAN
HI When in SRST either SRST Proper or CME as SRST ,when i call from HQ to branch office do i need to assign unregistered DN in order to have ext to ext calling working case of WAN fails ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] CCIE Voice LAB Rental rack in Gulf
Hi, I have my personal rack for ccie voice lab practice, if you want we can share and it will be cheapest, you can contact me on ccievoicedub...@gmail.com or 00973-33457553 Thanks Mick ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Voice Rack
Hi, I have the required devices to study CCIE LAB v.3, it does not match the requirements exactly but it is very good. To save my time I need to upload the initial configuration for each device at the beginning of each lab. I need to build an automated way to do this rather than do everything manually, like IP-Expert rack, anybody can help me to do this and suggest the best solutions for this? Regards, ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CCIE Payment
Thank you all, I contacted Cert. Support and I am waiting there reply. Regards, From: Ashraf Ayyash ash.ayy...@gmail.com To: Ken Wyan kew...@gmail.com Cc: CCIE Study ccie_voice@onlinestudylist.com Sent: Friday, November 25, 2011 8:43 PM Subject: Re: [OSL | CCIE_Voice] CCIE Payment its not a rule , it depend on what do you work as in Cisco and on the Center you are working with . Ash On Fri, Nov 25, 2011 at 10:49 AM, Ken Wyan kew...@gmail.com wrote: As I know , Cisco employees could give CCIE Lab Exams free of Exam Cost for 2~3 number of attempts. Is this facility still available? On Fri, Nov 25, 2011 at 1:50 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote: Go Ahead and contact the Certification Support to track this down , how long has it been since you applied the exam ? I had similar issue when i took the exam and the certification support team sort it out for me Best of Luck Ash On Thu, Nov 24, 2011 at 9:16 PM, Ccie Voice v.c...@yahoo.com wrote: Hi all, I added my credit card to pay for Cisco but they did not proceed the payment. I did not take care about it, because someone told me that Cisco now proceeding the payment after the lab, I went for CCIE lab and I did my lab but till now I did not receive my result and the payment still pending. Any body attempt soon can tell me if he paid before or after the lab? Regards, ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] WAN QoS
I am not sure also. but can we use auto qos voip trust? From: datucha123 datucha123 datucha...@gmail.com To: Ccie Voice v.c...@yahoo.com Cc: CCIE Study ccie_voice@onlinestudylist.com Sent: Monday, November 21, 2011 1:19 PM Subject: Re: [OSL | CCIE_Voice] WAN QoS As I know, we have to calculate the FRF.12 size based on the Access Rate, and not per the PVC Speed. So in this case we have to find out the actuall Access Rate of the FR link. So we must NOT calculate the FRF.12 based on the 384, but use the actuall FR Access Rate. Am I right? or not? On Mon, Nov 21, 2011 at 10:51 AM, Ccie Voice v.c...@yahoo.com wrote: Hi all, Could you please help me to solve this: There is 384 frame-rely PVC between HQ and BR1. Enable FRF.12 on this circuit , 10 ms as sampling rate. 16k for signaling traffic and 4 g.729 calls. Assume all RTP traffic is marked with EF and signaling with CS3 Header Compression should be enabled Configure LLQ and guarantee signaling get 16 K Thanks in advanced. ___ 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] auto qos command does not show up
added it under frame-relay interface-dlci 202 let me know if it is OK? From: Raees Shaikh racerra...@yahoo.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Sunday, November 20, 2011 9:18 AM Subject: [OSL | CCIE_Voice] auto qos command does not show up From: Raees Shaikh racerra...@yahoo.com To: CCIE Study ccie_voice@onlinestudylist.com Sent: Sunday, November 20, 2011 5:15 PM Subject: auto qos command does not show up Hi All, Below is the config from my lab HQ router ! nterface Serial0/3/0 no ip address encapsulation frame-relay clock rate 64000 frame-relay intf-type dce interface Serial0/3/0.1 point-to-point bandwidth 768 ip address 10.10.111.1 255.255.255.0 ip pim sparse-dense-mode frame-relay interface-dlci 201 interface Serial0/3/0.2 point-to-point ip address 10.10.112.1 255.255.255.0 ip pim sparse-dense-mode frame-relay interface-dlci 202 ! Im unable to configure auto qos as the command does not show up HQ-RTR(config)#interface Serial0/3/0.1 point-to-point HQ-RTR(config-subif)# bandwidth 768 HQ-RTR(config-subif)#aut HQ-RTR(config-subif)#auto HQ-RTR(config-subif)#auto? % Unrecognized command HQ-RTR(config-subif)#auto ? % Unrecognized command Am I missing something? HQ-RTR#sh ver Cisco IOS Software, 3800 Software (C3825-ADVENTERPRISEK9_IVS_LI-M), Version 12.4(24)T2, RELEASE SOFTWARE (fc2) TR ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] WAN QoS
Hi all, Could you please help me to solve this: There is 384 frame-rely PVC between HQ and BR1. Enable FRF.12 on this circuit , 10 ms as sampling rate. 16k for signaling traffic and 4 g.729 calls. Assume all RTP traffic is marked with EF and signaling with CS3 Header Compression should be enabled Configure LLQ and guarantee signaling get 16 K Thanks in advanced.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] SRST Advanced Config
Hi all, is it possible to configure: 1- Maximum Number of Calls 2- Busy Trigger In SRST Call-Manger-fallback NOT CME SRST? ___ 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 Advanced Config
Thank you Ashraf for your reply, but could you please help more. if I need to configure the following: Maximum Number of Calls: 4 Busy Trigger 2 how I can configure the above using huntstop channel command? Thanks in advanced. From: Ashraf Ayyash ash.ayy...@gmail.com To: Ccie Voice v.c...@yahoo.com Cc: CCIE Study ccie_voice@onlinestudylist.com Sent: Sunday, November 20, 2011 1:20 AM Subject: Re: [OSL | CCIE_Voice] SRST Advanced Config you can use huntstop channel command under the call manager fall back which will limit both in and out number of the calls on the dual/octo lines Ash On Sat, Nov 19, 2011 at 9:24 PM, Ccie Voice v.c...@yahoo.com wrote: Hi all, is it possible to configure: 1- Maximum Number of Calls 2- Busy Trigger In SRST Call-Manger-fallback NOT CME SRST? ___ 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 Advanced Config
thank you for your reply, but I did not understand exactly what I should configure? Could you please explain to me how huntstop channel can solve this? and what I should use huntstop cahnnel 1, 2 .. ? Regards, From: Ashraf Ayyash ash.ayy...@gmail.com To: Ccie Voice v.c...@yahoo.com Cc: CCIE Study ccie_voice@onlinestudylist.com Sent: Sunday, November 20, 2011 2:35 AM Subject: Re: [OSL | CCIE_Voice] SRST Advanced Config Hello , you cannot do this with call manager fallback , the hunt stop channel is global command for all the phones and for in/outbound calls on the lines . Ash On Sat, Nov 19, 2011 at 11:13 PM, Ccie Voice v.c...@yahoo.com wrote: Thank you Ashraf for your reply, but could you please help more. if I need to configure the following: Maximum Number of Calls: 4 Busy Trigger 2 how I can configure the above using huntstop channel command? Thanks in advanced. From: Ashraf Ayyash ash.ayy...@gmail.com To: Ccie Voice v.c...@yahoo.com Cc: CCIE Study ccie_voice@onlinestudylist.com Sent: Sunday, November 20, 2011 1:20 AM Subject: Re: [OSL | CCIE_Voice] SRST Advanced Config you can use huntstop channel command under the call manager fall back which will limit both in and out number of the calls on the dual/octo lines Ash On Sat, Nov 19, 2011 at 9:24 PM, Ccie Voice v.c...@yahoo.com wrote: Hi all, is it possible to configure: 1- Maximum Number of Calls 2- Busy Trigger In SRST Call-Manger-fallback NOT CME SRST? ___ 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] Call Routing
Hi Chris, thank you for your help, yes you are right I should not use ^ in destination pattern now it looks OK but I do know what was the exact problem. yesterday I did many changes. but before if I used debug isdn q931 I will not get any output. but if I used debug voip ccapi I can see that I am sending the exact digits and I am hitting the exact dial-peer but no call routing. and I checked the disconnect cause, it was 1 = unassigned number. I will try to rebuild the lab soon and check again :) From: Chris Martin clm.c...@gmail.com To: Ccie Voice v.c...@yahoo.com Cc: CCIE Study ccie_voice@onlinestudylist.com Sent: Saturday, October 22, 2011 4:34 PM Subject: Re: [OSL | CCIE_Voice] Call Routing If you add a pots dialpeer with an ^ to explicitly state the beginning of the match pattern, then it doesn't follow the rules for stripping the digits it matches. You are not doing anything wrong, just a weird exception to the rule. You can confirm this with a debug isdn q931 and see the 9 being sent to the pstn. HTH, Chris On Fri, Oct 21, 2011 at 5:59 AM, Ccie Voice v.c...@yahoo.com wrote: Hi all, I have very strange problem, and I need someone to help me to understand why? I am trying to study call routing, local calls I have the following setup SCCP Phone RP Local RL H.323 GW PSTN in the GW I added the following dial-peer: dial-peer voice 15 pots translation-profile outgoing loc destination-pattern ^9[2-9]..$ port 0/1/0:23 while the dial-peer is pots so the 9 should be stripped and remaining will be send to pstn but the call is not working unless I added forward-digits 7 anybody can help me to understand why?? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Voice Gateway and Signaling
Hi everybody, If a question asked you to configure the voice gateways as the following: use T1 Framing: ESF, line coding :B8ZS isdn sw: primary-ni take clocking for layer 1 from network side your pri circuit layer 2 should be user side for me the answer will be the following configuration: isdn switch-type primary-ni network-clock-participate wic 1 network-clock-select 1 T1 0/1/0 controller T1 0/1/0 cablelength long 0db pri-group timeslots 1-4,24 service mgcp ! interface Serial0/1/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn bind-l3 ccm-manager isdn outgoing display-ie isdn outgoing ie redirecting-number no cdp enable ! Is there anything wrong in my config?? Please if somebody know any trick in this section please help me.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Call Routing
Hi all, I have very strange problem, and I need someone to help me to understand why? I am trying to study call routing, local calls I have the following setup SCCP Phone RP Local RL H.323 GW PSTN in the GW I added the following dial-peer: dial-peer voice 15 pots translation-profile outgoing loc destination-pattern ^9[2-9]..$ port 0/1/0:23 while the dial-peer is pots so the 9 should be stripped and remaining will be send to pstn but the call is not working unless I added forward-digits 7 anybody can help me to understand why?? ___ 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] Call Routing
Hi all and thank you to answer my question, but still I am not able to solve it without forward digits command: I tried to remove ^ but when I removed it and tried to make calls I hear dial-tone from PSTN router. And regarding : The gateway/dial-peer takes the [2-9] as a pattern match, also and strips them. If it is like this why when I used debug isdn q931 on pstn router I found no called party number ?? if it is considering [2-9] as a pattern match then I should find in pstn that the remaining as called number. Am I right? Regards, From: Raees Shaikh racerra...@yahoo.com To: Ccie Voice v.c...@yahoo.com; CCIE Study ccie_voice@onlinestudylist.com Sent: Friday, October 21, 2011 6:21 PM Subject: Re: [OSL | CCIE_Voice] Call Routing Hi What digits are going when you dont add forward digits 7? Regards, Raees From: Ccie Voice v.c...@yahoo.com To: CCIE Study ccie_voice@onlinestudylist.com Sent: Friday, October 21, 2011 4:29 PM Subject: [OSL | CCIE_Voice] Call Routing Hi all, I have very strange problem, and I need someone to help me to understand why? I am trying to study call routing, local calls I have the following setup SCCP Phone RP Local RL H.323 GW PSTN in the GW I added the following dial-peer: dial-peer voice 15 pots translation-profile outgoing loc destination-pattern ^9[2-9]..$ port 0/1/0:23 while the dial-peer is pots so the 9 should be stripped and remaining will be send to pstn but the call is not working unless I added forward-digits 7 anybody can help me to understand why?? ___ 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] Voice Gateway and Signaling
Thank you but I do not think I am correct :) From: Robert Schuknecht rschukne...@gmx.de To: 'Ccie Voice' v.c...@yahoo.com; 'CCIE Study' ccie_voice@onlinestudylist.com Sent: Friday, October 21, 2011 4:50 PM Subject: AW: [OSL | CCIE_Voice] Voice Gateway and Signaling I think you are correct. /Robert Von:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] Im Auftrag von Ccie Voice Gesendet: Freitag, 21. Oktober 2011 12:01 An: CCIE Study Betreff: [OSL | CCIE_Voice] Voice Gateway and Signaling Hi everybody, If a question asked you to configure the voice gateways as the following: use T1 Framing: ESF, line coding :B8ZS isdn sw: primary-ni take clocking for layer 1 from network side your pri circuit layer 2 should be user side for me the answer will be the following configuration: isdn switch-type primary-ni network-clock-participate wic 1 network-clock-select 1 T1 0/1/0 controller T1 0/1/0 cablelength long 0db pri-group timeslots 1-4,24 service mgcp ! interface Serial0/1/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn bind-l3 ccm-manager isdn outgoing display-ie isdn outgoing ie redirecting-number no cdp enable ! Is there anything wrong in my config?? Please if somebody know any trick in this section please help me.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] MOH multicast
Hi everybody, I am trying to configure Multicast MOH in Branch 1 router, I added the file to router flash and I configure CUCM to use multicast. when I am calling from one phone to another in BR1 I am able to hear the file that I uploaded to router flash memory. but if I tried to use the following command: sho ccm-manager music-on-hold the result: Current active multicast sessions : 0 is it using Multicast? how I can make sure that I am using multicast not unicast? ___ 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] MOH multicast
But I am using router flash memory Not CUCM___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Locations vs. RSVP CAC
Hi everybody, When I have to use CAC location based and when to use CAC RSVP? - Multi-link, are there any other reasons or scenarios? What is the meaning for the following? When using locations in hub-and-spoke topology, devices at the hub site should assigned to the None Location From CCIE Voice Exam Quick Reference sheets Page: 61___ 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] Locations vs. RSVP CAC
Thank you Ken, Yes I agree with you but what I need to know exactly, what is the advantage and disadvantage of all methods or when it is recommended to use RSVP and when it is recommended or it is better to use location based CAC. From: Ken Wyan kew...@gmail.com To: Ccie Voice v.c...@yahoo.com Cc: CCIE Study ccie_voice@onlinestudylist.com Sent: Wednesday, September 28, 2011 1:05 PM Subject: Re: [OSL | CCIE_Voice] Locations vs. RSVP CAC Locations based CAC has a total value (total calls in out of) per each location. Hence not suitable for Hub. RSVP has more realistic control , it's topology aware , but need to configure each router in the path register each as a MTP in CUCM On Wed, Sep 28, 2011 at 10:30 AM, Ccie Voice v.c...@yahoo.com wrote: Hi everybody, When I have to use CAC location based and when to use CAC RSVP? - Multi-link, are there any other reasons or scenarios? What is the meaning for the following? When using locations in hub-and-spoke topology, devices at the hub site should assigned to the None Location From CCIE Voice Exam Quick Reference sheets Page: 61 ___ 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] TCS capabilities
I did this before, you can use debug cch323 h225 debug cch323 h245 after Ringing you will find the party under waiting h245 capabilities, I recommend to read SRND it has some good information about this. Regards, From: DeShon Crayton dcrayto...@comcast.net To: 'Nowork_onlyfun' noworkonlyf...@gmail.com; ccie_voice@onlinestudylist.com Sent: Wednesday, September 28, 2011 5:06 PM Subject: Re: [OSL | CCIE_Voice] TCS capabilities TCS basically works when there are PSTN TDM connections. In the voice lab, I would stick to not checking the TCS box. Depending on the lab and specific requirements, you are more than likely to need a codec specific MTP. With cube you will probably need MTP registered to the router. Strait gatekeeper will more than likely require a MTP registered to UCM. Debug h225 asn1 is a good command Debug ras is another good command -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Nowork_onlyfun Sent: Wednesday, September 28, 2011 6:43 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] TCS capabilities Hello guys. I am doing a lab where my gatekeeper is cube as well. When I ring pstn number 0091! The cucm phones shows connected and pstn phone still ringing. To fix it I uncheck wait for TCS far end capabilities. That seems to fix the issue because cube is involved in this scenario. Which debug should I run to see this or how can I check from debugs that I need to uncheck that ? Thanks a lot. Sent from my iPad ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Router for My Home LAB
Hi all, I need to add a router for my home lab and I need your advise about which one I should choose. I know that the best one is 2801 but it is very expensive if I need to compare with other old routers. what I know, I can use one of the following routers: 3725 2620MX 1760 3640 ( I think this router it is not good to use in the lab and there is no 12.4 IOS , the same one used in the lab) Please help me to choose the best one, as I said before 2801 is the best but it it expensive. but at the end I need a router which allow me to do all CCIE tricks. Regards, ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Second Attempt
Hi All, I did my first attempt, I did not pass but the report was very good and most of my marks 100%. The report encourages me to do my second attempt. I prepared for the second attempt and I did my second attempt. Second attempt was much better than the first one. I studied very well and tried to avoid my mistakes in my first attempt. Unfortunately, the report was very bad. Most of my marks 40%, 20%, and some 0%. But why??? Some questions are exactly the same with first attempt? Why I am getting 100 in my first attempt and 0 in my second attempt. Some of the requirements I was not able to solve it and thought I will get 0 because I did not solve it but in the report 100%. Is it a magic? I am now very pessimistic I am thinking to leave my CCIE and I am sometimes I thinking to leave Cisco and work with other vendors. I am very sorry for this bad e-mail. Specially, for who is preparing right now. But I am really very pessimistic from my second attempt and I need some advises from your side Regards,___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Logistic info for the lab
Hi Vega, 1. What can we bring in to the lab? Water bottle? No thing :) 2. How long is the lunch break? Is it allocated at a specific time? it is 30 mins the proctor should tell you about it. 3. Do we have to do the patching of cables? No 4. What do they provide apart from the Cisco PDF? Any writing papers? you will not get Cisco PDF you should have access to Cisco support and documentation site and they will provide you with two or 3 papers. From: Vega Wong vega2...@yahoo.com.au To: ccie_voice@onlinestudylist.com Sent: Wed, July 13, 2011 5:36:06 PM Subject: [OSL | CCIE_Voice] Logistic info for the lab Hi experts Many experts had advised that time management is a very important ( if not the most important) aspect during the lab. This leads me to wonder some of the smaller things during the lab: 1. What can we bring in to the lab? Water bottle? 2. How long is the lunch break? Is it allocated at a specific time? 3. Do we have to do the patching of cables? 4. What do they provide apart from the Cisco PDF? Any writing papers? Obviously I havent had my first attempt yet, but I will have mine soon. Just thought the more I know the better, Hope those who had the experience can share Cheers Vega___ 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] PSTN-WAN call drop scenario
Dear Friends, I have created a group on Skype where we can communicate online with each other to discuss our matters on urgent basis, you guys can add ( ccie.voice.group ) where you can be added to the group. Just copy paste this request while adding ccie.voice.group ( I am a Cisco Voice User, and would like to be added to the group. I am from Country working for Company and have been in this field since number of years, my email address is email address) Regards, On Mon, Jul 11, 2011 at 12:47 AM, Shabeeb Mohammed shabeebc...@gmail.comwrote: Thanks Santiago.. I went through the document.. there is no wrong in what i have done.. damn.. this is confusing.. :( On Sun, Jul 10, 2011 at 5:49 PM, Santiago Figueroa sfigue...@mnet.com.mxwrote: ** I have this information for you. ** ** ** ** -- *De:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *En nombre de *Shabeeb Mohammed *Enviado el:* Sábado, 09 de Julio de 2011 11:33 p.m. *Para:* **ccie_voice@onlinestudylist.com** *Asunto:* Re: [OSL | CCIE_Voice] PSTN-WAN call drop scenario ** ** Dear All, ** ** Im trying to integrate CUCM 8 with the AD. I have created a user ccmseeb under Seeb*-*R E*-*Users... please see the first attached file. Now im trying to integrate with the CUCM. and no matter what ever i do, it just will not work. The * LDAP User Search Base* i have created is OU=Users,OU=R E,OU=Seeb,DC=***,DC=***,DC=*** (ive just omitted the DC details.. i assure you that it is correct) The AD is reachable from the cucm. But when i Click *Save* (under LDAP Directory), i get the following * error* Wrong LDAP User Search Base OU=Users,OU=R E,OU=Seeb,DC=***,DC=***,DC=*** while Connecting to ldap://172.16.39.21:389 please see the 2nd attached file What could be the problem.. anyone??? ** ** Also I know that this is not CCIE kind of hard, but ive been stuck with this for a while now.. ** ** Thanks Shabeeb Mohammed ** ** ** ** On Thu, Jul 7, 2011 at 8:59 PM, George Goglidze gogli...@gmail.com wrote: Hi Adil, ** ** Without debug h245 asn1 it's difficult to be sure, but I bet on this other rack rental they do support g729 codec on incoming dial-peer on PSTN-WAN router and thats why the call succeeds. Sent from my iPad On 7 Jul 2011, at 02:08, Adil Shaikh adil.sha...@gmail.com wrote: hi experts, in ipexpert lab, i make call from HQ phone to PSTN-WAN via CUBE, when the call is answered by PSTN-WAN phone, call fails. This is due to codec mismatch and i can get 'TSC rejected' message in 'debug h245 asn1'. I have been succeful in simulating this on everytime in last 5 attempts on ipexpert lab. few days back, i used other providers lab for which i paid lot of money. i did same configuration as i do on ipexpert lab (that is create g729 region which talks to HQ and BR1 region as g729. Assign g729 region to GK device pool. Assign this device pool to GK trunk). (note: I use this trunk to make a 4 digit call to BR2 phone and when i do 'show gatekeeper call', i see 16k used. So, i know that GK trunk is definately using g729 codec.) for PSTN-WAN, i configured CUBE as well. When i make a call to PSTN-WAN when the call is answered call does not drop. I was expecting it to drop due to codec mismatch. Here is my configuration on HQ router: voice service voip all h t h gatkeeper zone local VIA http://ipexpert.comipexpert.com zone local GK http://ipexpert.comipexpert.com 10.10.110.1 using local loopback zone remote PSTN-WAN http://joblog.comjoblog.com outvia VIA zone prefix PSTN-WAN 01191* zone prefix GK 212* gw-priority 10 gk-trunk_2 zone prefix GK 212* gw-priority 9 gk-trunk_1 zone prefix GK 617* gw-priority 10 gk-trunk_2 zone prefix GK 617* gw-priority 9 gk-trunk_1 zone prefix GK 212* gw-priority 0 CME zone prefix GK 617* gw-priority 0 CME no shut int lo 0 ip add 10.10.110.1 255.255.255.0 h323-g v b s 10.10.110.1 h323-g v inter h323-g v id VIA ipadd 10.10.110.1 h323-g v h323-id HQ-RTR gateway dial-p voice 91 voip destination-p 01191T incoming called 01191 session target ras dtmf-r h245-al no vad On CUCM: a route pattern 9.011916737485173 partition: pt-hq-CallToIndia discard digit predot routelist: rl-gk-CallToIndia contains gk trunk calling party External number mask: On Do you guys see any problem in above config. Have you come across similar situation in non-IPexpert lab for which you have paid lot of money? If so, how did you resolve
Re: [OSL | CCIE_Voice] B-acd - Not working
Dear Friends, I have created a group on Skype where we can communicate online with each other to discuss our matters on urgent basis, you guys can add ( ccie.voice.group ) where you can be added to the group. Just copy paste this request while adding ccie.voice.group ( I am a Cisco Voice User, and would like to be added to the group. I am from Country working for Company and have been in this field since number of years, my email address is email address) Regards, On Sun, Jul 10, 2011 at 11:43 PM, ccieid1ot ccieid...@gmail.com wrote: Try binding your cme to your vlan31. Also, try add a hunt-group. Just taking a stab in the dark. On Jul 10, 2011 2:43 AM, Sanoj Thomas san...@yahoo.com wrote: ___ 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] CCX 7.0.2 and CUCM 7.1.5 JTAPI issue Client issue
Dear Friends, I have created a group on Skype where we can communicate online with each other to discuss our matters on urgent basis, you guys can add ( ccie.voice.group ) where you can be added to the group. Just copy paste this request while adding ccie.voice.group ( I am a Cisco Voice User, and would like to be added to the group. I am from Country working for Company and have been in this field since number of years, my email address is email address) Regards, On Fri, Jul 8, 2011 at 2:32 AM, Thomas Koch koch1...@comcast.net wrote: Team, Anyone have any issues with CCX 7.0.2 and CUCM 7.1.5? I’m having JTAPI issues..the matrix from Cisco says it should work…no joy…The CCX Engine is in partial service.. Thoughts? ** ** ** ** Thomas Koch Owner/Consultant CCDA, CCNA, CCNP Voice Cisco IPT Design Specalist Digitones, LLC Cell: +1.630.235.4309 E-mail: digito...@comcast.net ___ 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] SIP auto-registration failure with CIPC
Dear Friends, I have created a group on Skype where we can communicate online with each other to discuss our matters on urgent basis, you guys can add ( ccie.voice.group ) where you can be added to the group. Just copy paste this request while adding ccie.voice.group ( I am a Cisco Voice User, and would like to be added to the group. I am from Country working for Company and have been in this field since number of years, my email address is email address) Regards, 2011/7/9 khaled Saholy khaled_sah...@hotmail.com Hi , I have a problem when I tried to use sip auto-registration in cucm 7.0 to register CIPC IPPhone as SIP endpoint. I changed the settings in the Enterprise parameters to SIP , restarted call manager service. I even rebooted the call manager but still the same. It got registered but as SCCP. Any idea about this problem. Regards. Khaled ___ 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] Couldn't browse script/prompt repository
Dear voice Users, I have created a group on Skype where we can communicate online with each other to discuss our matters on urgent basis, you guys can add ( ccie.voice.group ) where you can be added to the group. Just copy paste this request while adding ccie.voice.group ( I am a Cisco Voice User, and would like to be added to the group. I am from Country working for Company and have been in this field since number of years, my email address is email address) At the same time you can carry on with communicating on this group. Regards, On Tue, Jul 5, 2011 at 8:03 AM, Alex Goh ncsalex@gmail.com wrote: Hi Ron, Yes I already have an application using the default ICD script, but still I can't browse or save on the editor's repository Hi Santiago, I'm log in with uccxadmin account. I know anonymous account will not be able to run reactive script and browse/save script repository. Regards, Alex On Mon, Jul 4, 2011 at 11:53 PM, Santiago Figueroa sfigue...@mnet.com.mxwrote: ** When you logg in CCX editor to do that uccxadmin? ** ** -- *De:* **ccie_voice-boun...@onlinestudylist.com** [mailto:** ccie_voice-boun...@onlinestudylist.com**] *En nombre de *Alex Goh *Enviado el:* Lunes, 04 de Julio de 2011 09:32 a.m. *Para:* OSL *Asunto:* [OSL | CCIE_Voice] Couldn't browse script/prompt repository ** ** Hi Guys, I'm having a weird issue with my UCCX 7.0 (1) Build 109. When login to the CCX editor, I'm not able to save or browse the script/prompt by clicking the repository button (see attached). I've tried reinstalled my UCCX with the repair options, rebooted my UCCX server, created a new uccx admin, and tried running CCX editor from another box all no lucks. Also, I notice the UCCX only works on default script, whenever I save as a default script like icd.aef without modifying it to the script/system/default folder. the UCCX will turn into partial service state,. where Application Manager is the one that OOS. By the way, the uccxadmin end user account was assigned with CCM Super User and allow CTI control all group, if this is related. Can someone shed some light on this? Regards, Alex -- La información incluida en este mensaje y sus anexos es CONFIDENCIAL y para USO EXCLUSIVO de sus destinatarios. No está permitida su divulgación y/o reproducción sin autorización. Si ha recibido este mensaje y no le incumbe, le rogamos nos los comunique y proceda a su borrado. Gracias. Information included in this e-mail and attached files is CONFIDENTIAL and only for the EXCLUSIVE USE of the receivers. Circulation and/or copy without permission is not allowed. If you have received this e-mail and you are not the intended recipient, please let us know and erase the message and attached files. Thank you. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Skype Group
Dear voice Users, I have created a group on Skype where we can communicate online with each other to discuss our matters on urgent basis, you guys can add ( ccie.voice.group ) where you can be added to the group. Just copy paste this request while adding ccie.voice.group ( I am a Cisco Voice User, and would like to be added to the group. I am from Country for Company and working in this field since number of years ). Also recommend it to the people related in this field to your connections. At the same time you can carry on with communicating on this group as well. Regards, ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] How to avoid phone firmware upgrade in the lab
Do you say that Lab will not be restored to a pre-decided configurations on all devices before begining of each test? OR in initial lab setup , phones not having access to CUCM / CME for firmware changes? Dave On Wed, Jun 22, 2011 at 2:21 AM, Roger Carpio roger.car...@gmail.comwrote: If the previous candidate using your lab spot changed the protocol as requested, let's say his lab had SIP and now yours is SCCP; there is nothing you can do to avoid this. It is not necessarily a waste of time; unless you keep staring at the phones LOL Regards, Roger Carpio. On Tue, Jun 21, 2011 at 2:16 PM, Pithog Oil pithog...@yahoo.com wrote: Hi Experts, what should i do so as to avoid phone firmware upgrade at the lab, i have discovered that some times i unknowingly have a phone firmware upgrade from one version to another. Kind assist it is usually a waste of time. Pithog Oil ___ 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 http://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 http://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
