[OSL | CCIE_Voice] Ok...what's theTFTP service called for CLI restart
utils service restart Cisco Tftp Thanks, Rohith ___ 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] 5 new labs lab 1 - cue
looking at it over I think it was i did not include: voice service voip allow-connections h323 to sip etc etc etc Cheers, Randall From: Vik Malhi vma...@ipexpert.com To: Randall Crumm rrcr...@yahoo.com Cc: Online Study ccie_voice@onlinestudylist.com Sent: Saturday, June 9, 2012 8:42 PM Subject: Re: [OSL | CCIE_Voice] 5 new labs lab 1 - cue The troubleshooting methodology has to be to eliminate various items involved in the call. Change region between br2 and HQ to g711- does that work? Can you call direct to the cue pilot number from HQ/br1 or is this isolated to call forward? Can you call the cue pilot from br2 phones? Remove any location cac to eliminate an ouf bandwidth flag Does the cue have an mrgl to see the Xcoder at br2? Is the Xcoder in right dpool? Are you using RSVP? Is there above an overlapping codec in the mtp and Xcoder? -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 On Jun 9, 2012, at 12:04, Randall Crumm rrcr...@yahoo.com wrote: Hi, I am starting up again. I am trying to leave a message for branch 2 user. Branch 2 has CUE. I configured CTI PR, CTI ports,VM pilot,applied VM pilot to br2ph2 line. configured transcoder on br2 gw, registered br2 transcoder to CUCM. applied transcoder to a MRG When I call from br1 I get a fast busy Any thoughts? Cheers, Randall ___ 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] Blueprint Updates
Hello Vik, Good to know we still have some time to go with the V3 blueprint AWOJOODU FUNSO 08033344658 Don't worry so much about tomorrow that you forget to live today. From: ccie_voice-requ...@onlinestudylist.com ccie_voice-requ...@onlinestudylist.com To: ccie_voice@onlinestudylist.com Sent: Monday, 11 June 2012, 0:35 Subject: CCIE_Voice Digest, Vol 76, Issue 26 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. Blueprint update (Vik Malhi) 2. Re: Last Chance to Register for IPexpert?s Online Voice ?Alchemy? Class?. (Rrcrumm) 3. Re: Blueprint update (Amp) 4. callback (Joel Petralia) 5. Requested circuit/channel not available (chase mergenthal) -- Message: 1 Date: Sun, 10 Jun 2012 12:23:04 -0700 From: Vik Malhi vma...@ipexpert.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Blueprint update Message-ID: a2229c97-456f-4d20-8c3a-c58626f67...@ipexpert.com Content-Type: text/plain; charset=utf-8 The message from CIsco Live is Dont expect an announcement any time soon re: the blueprint change for CCIE-V. My interpretation is this means you have the rest of this year and fairly deep into 2013 before a blueprint change. Understandably not much information is being given but I will share my thoughts on our blog next week. If you are mid way through your studies then you have ample time- I would encourage you to not get distracted and get it done on this version. -- Vik Malhi ? CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Wireless, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ -- Message: 2 Date: Sun, 10 Jun 2012 14:13:27 -0700 From: Rrcrumm rrcr...@yahoo.com To: Vik Malhi vma...@ipexpert.com Cc: OSL ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Last Chance to Register for IPexpert?s Online Voice ?Alchemy? Class?. Message-ID: 09bf393c-2e09-47c8-bc64-ffbb085d3...@yahoo.com Content-Type: text/plain; charset=utf-8 Lol Vik You are the Man!!! Randall Sent from my iPhone On Jun 10, 2012, at 12:12 PM, Vik Malhi vma...@ipexpert.com wrote: Yes I can confirm I am alive and well and am at IPX. I was worried there for a while that you guys on the list had heard something so it is a relief to hear Wayne say that:-) -- Vik Malhi ? CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Wireless, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ On Jun 9, 2012, at 8:56 PM, Wayne Lawson waynelawson-...@ipexpert.com wrote: Vik is definitely @ IPexpert. He's an officer and shareholder and isn't going anywhere! ;-) Regards, Wayne A. Lawson II - CCIE #5244 Founder President IPexpert, Inc., Proctor Labs, Inc., Masonic e-Institute of Technology, Inc., Platinum, Inc. Mobile: +1.810.334.1564 eFax: +1.810.454.0130 Email: wlaw...@ipexpert.com Connect @ www.WayneLawson.com :: Message sent from iPhone. On Jun 9, 2012, at 4:43 PM, donny f f.faraday...@gmail.com wrote: Hi all, Is vik not with ipx anymore? Sorry not following list for while. D On Saturday, June 9, 2012, Bill Lake whl...@gmail.com wrote: Kevin Wallace and Anthony Sequeira are both very good and
Re: [OSL | CCIE_Voice] no trace found but calls get routed br1-Hq
Hi folks, I couldn't understand the call flow between HQ and BR1 which are provisioned/registed in the cucm. here is the detail structure: HQ-phone1 -5002 css-hq-international pt-pt-internal BR1-phone1-1002 css-br1-ld pt-pt-internal Both phones are residing in the partition pt-internal, and br1 is a mgcp site and whereas the hq is the h323 site. when i call 1002 from 5002 or vice versa the call works fine, but when i enable deb isdn q931 or deb voip dialp, i dont see anything. Whereas when i enable RSVP based CAC, i can see the traces with the show sccp connections. could any one help me out how the calls are working in between these two. is it because the phones are registered to cucm, but logically in a different device pool and therefore it routes directly on cucm your help is much appreciated. Thank you. Krishna.___ 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] 5 new labs lab 1 - cue
You don't need allow connections since it is a cti integration not sip. -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 On Jun 10, 2012, at 22:14, Randall Crumm rrcr...@yahoo.com wrote: looking at it over I think it was i did not include: voice service voip allow-connections h323 to sip etc etc etc Cheers, Randall From: Vik Malhi vma...@ipexpert.com To: Randall Crumm rrcr...@yahoo.com Cc: Online Study ccie_voice@onlinestudylist.com Sent: Saturday, June 9, 2012 8:42 PM Subject: Re: [OSL | CCIE_Voice] 5 new labs lab 1 - cue The troubleshooting methodology has to be to eliminate various items involved in the call. Change region between br2 and HQ to g711- does that work? Can you call direct to the cue pilot number from HQ/br1 or is this isolated to call forward? Can you call the cue pilot from br2 phones? Remove any location cac to eliminate an ouf bandwidth flag Does the cue have an mrgl to see the Xcoder at br2? Is the Xcoder in right dpool? Are you using RSVP? Is there above an overlapping codec in the mtp and Xcoder? -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 On Jun 9, 2012, at 12:04, Randall Crumm rrcr...@yahoo.com wrote: Hi, I am starting up again. I am trying to leave a message for branch 2 user. Branch 2 has CUE. I configured CTI PR, CTI ports,VM pilot,applied VM pilot to br2ph2 line. configured transcoder on br2 gw, registered br2 transcoder to CUCM. applied transcoder to a MRG When I call from br1 I get a fast busy Any thoughts? Cheers, Randall ___ 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] UCCX - Do CTI Ports in Partition work ?
With UCCX, did anybody get calls to work when the CTI ports are in a partition ? If so what CSS did you have to configure ? I have created a RoutePoint in the NULL Partition and CTI ports in a UCCX partition Added a CSS for the RoutePoint that includes the UCCX partition (on both the line device) but the call doesn't connect. If i take the CTI ports out of the partition, everything works perfectly. TIA -- - Pavan ___ 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] no trace found but calls get routed br1-Hq
You answered your own question. Both DNs are registered to CUCM and are in partitions that the other's CSS can see. The signaling is between each phone and UCM. The media is built directly phone to phone. If CAC failed the call setup, then AAR could be invoked to use the PSTN. Since CAC allows the call, the gateways aren't involved in the call (other than providing IP network connectivity.) DQ d...@cisco.com Sent from my iPhone On Jun 11, 2012, at 2:52 PM, Krishna vinayak_...@yahoo.com wrote: Hi folks, I couldn't understand the call flow between HQ and BR1 which are provisioned/registed in the cucm. here is the detail structure: HQ-phone1 -5002 css-hq-international pt-pt-internal BR1-phone1-1002 css-br1-ld pt-pt-internal Both phones are residing in the partition pt-internal, and br1 is a mgcp site and whereas the hq is the h323 site. when i call 1002 from 5002 or vice versa the call works fine, but when i enable deb isdn q931 or deb voip dialp, i dont see anything. Whereas when i enable RSVP based CAC, i can see the traces with the show sccp connections. could any one help me out how the calls are working in between these two. is it because the phones are registered to cucm, but logically in a different device pool and therefore it routes directly on cucm your help is much appreciated. Thank you. Krishna. ___ 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] using own lab equipment (servers) andtroubleshooting exercises
You could create intentionally flawed base configs and use DRS to backup / restore them. Really though, you'll learn more by just making mistakes, troubleshooting your mistakes, and fixing those mistakes. DQ d...@cisco.com Sent from my iPhone On Jun 11, 2012, at 2:53 PM, Steve Nicklas steve.nickl...@gmail.com wrote: Hello all, When using your own servers, is it still possible to fully experience the troubleshooting sections in the labs? With IOS devices, of course it is easy to load up the intentionally flawed config file to the router to start troubleshooting. But with a CUCM for example, how can this be done? Or is this a concern? Thanks, Steve ___ 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] no trace found but calls get routed br1-Hq
Krishna, When dialing from station to station and both stations are registered to UCM, the call does not normally traverse through the PSTN (no AAR case). The signaling media flows over voip directly which is why you dont see any gateway / q931 debugs being active. However for a Voip flow to maintain proper quality, CAC/RSVP is used to ensure sufficient bandwidth being used which is why you see the RSVP debug active. Media flows from endpoint to endpoint directly through the RSVP agents which is what you see in sh sccp connections Signaling flows from endpoint to UCM direct. Remember the gateway is not in the signaling path which is why you do not see anything on the gw. On Mon, Jun 11, 2012 at 11:42 AM, Krishna vinayak_...@yahoo.com wrote: Hi folks, I couldn't understand the call flow between HQ and BR1 which are provisioned/registed in the cucm. here is the detail structure: HQ-phone1 -5002 css-hq-international pt-pt-internal BR1-phone1-1002 css-br1-ld pt-pt-internal Both phones are residing in the partition pt-internal, and br1 is a mgcp site and whereas the hq is the h323 site. when i call 1002 from 5002 or vice versa the call works fine, but when i enable deb isdn q931 or deb voip dialp, i dont see anything. Whereas when i enable RSVP based CAC, i can see the traces with the show sccp connections. could any one help me out how the calls are working in between these two. is it because the phones are registered to cucm, but logically in a different device pool and therefore it routes directly on cucm your help is much appreciated. Thank you. Krishna. ___ 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 -- - Pavan ___ 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 - Do CTI Ports in Partition work ?
Hi Pavan, We've seen this behavior with UCCX. Logically, the calls should work w/ or w/o partition applied on the CTI Port Group, keeping in mind the CSS applied on the CTI Route Point. Few things to keep in mind: 1) Always apply the changes on these Triggers/ Port Groups from the CCX and never from the CM. 2) If you apply the correct CSS on the Trigger which includes the partition of the Port group, the calls should work. 3) Even after applying the changes if the calls do not work, it could be very possible that the changes you're making from the CCX are not getting updated on the CM. In this case, first run the Data Resync from the CCX and make sure there are no exceptions in the output. Then, restart the CTI Manager on all CM servers and then restart the CCX Engine. - Gurpreet On Mon, Jun 11, 2012 at 2:33 PM, Pavan K pav.c...@gmail.com wrote: With UCCX, did anybody get calls to work when the CTI ports are in a partition ? If so what CSS did you have to configure ? I have created a RoutePoint in the NULL Partition and CTI ports in a UCCX partition Added a CSS for the RoutePoint that includes the UCCX partition (on both the line device) but the call doesn't connect. If i take the CTI ports out of the partition, everything works perfectly. TIA -- - Pavan ___ 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 - Do CTI Ports in Partition work ?
I would think that the inbound caller (ip phone or gw) would need the CSS to access the CTI ports. DQ d...@cisco.com Sent from my iPhone On Jun 11, 2012, at 4:09 PM, Pavan K pav.c...@gmail.com wrote: With UCCX, did anybody get calls to work when the CTI ports are in a partition ? If so what CSS did you have to configure ? I have created a RoutePoint in the NULL Partition and CTI ports in a UCCX partition Added a CSS for the RoutePoint that includes the UCCX partition (on both the line device) but the call doesn't connect. If i take the CTI ports out of the partition, everything works perfectly. TIA -- - Pavan ___ 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] no trace found but calls get routed br1-Hq
Thank you Dan for providing me the detail info. I assumed the same but not sure with my hypothesis. I am wondering if this is the case, then wan qos will not be able to do much isn't it... for instance yesterday i configured llq-cbfwq with bandwidth of 28 for rtp traffic between hq and br1. And, when i called the 5002 from 1002 the call went thru, and this call put on hold and placed another call and it works fine as well. So, from this analysis can i come to conclusion that only location based cac, or rsvp cac can only the number of calls between these two sites??? Thank you Krishna. From: Dan Quinlan (daquinla) daqui...@cisco.com To: Krishna vinayak_...@yahoo.com Cc: ccie_voice@onlinestudylist.com Sent: Monday, June 11, 2012 2:03 PM Subject: Re: [OSL | CCIE_Voice] no trace found but calls get routed br1-Hq You answered your own question. Both DNs are registered to CUCM and are in partitions that the other's CSS can see. The signaling is between each phone and UCM. The media is built directly phone to phone. If CAC failed the call setup, then AAR could be invoked to use the PSTN. Since CAC allows the call, the gateways aren't involved in the call (other than providing IP network connectivity.) DQ d...@cisco.com Sent from my iPhone On Jun 11, 2012, at 2:52 PM, Krishna vinayak_...@yahoo.com wrote: Hi folks, I couldn't understand the call flow between HQ and BR1 which are provisioned/registed in the cucm. here is the detail structure: HQ-phone1 -5002 css-hq-international pt-pt-internal BR1-phone1-1002 css-br1-ld pt-pt-internal Both phones are residing in the partition pt-internal, and br1 is a mgcp site and whereas the hq is the h323 site. when i call 1002 from 5002 or vice versa the call works fine, but when i enable deb isdn q931 or deb voip dialp, i dont see anything. Whereas when i enable RSVP based CAC, i can see the traces with the show sccp connections. could any one help me out how the calls are working in between these two. is it because the phones are registered to cucm, but logically in a different device pool and therefore it routes directly on cucm your help is much appreciated. Thank you. Krishna. ___ 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 76, Issue 32
I have been using my own equipment for the labs and practice and have not found this to be a detrimental issue. The only problem with it is this When you are looking at the initial configs to format them to work on your equipment , it's hard not to accidentally spot the purposefully entered configuration commands that create the troubleshooting aspect of the labs. Thus you must learn from those situations at different times than intended. As far as with the applications, I haven't really noticed anything that I would have done to create issues. UCCX issues are loaded for you in the Volume 2 labs so you can just load those up into your UCCX server to replicate the troubleshooting scenario. I haven't had any labs (that I can recall) that I would have had to create issues in the applications(UCM,CME,UCXn,CUE and CUPS) with. I hope I clearly wrote that. Anyways, my .02c is that you ultimately have to troubleshoot and account for more, thus gain a better understanding, when you use your own equipment. I believe this is cause you have to look more closely at the configs to replicate the environment developed by IPE onto your POD. There is give and take with this but I believe it is ultimately more complex and rewarding to use your own equipment. Just make sure you get the benefit of spotting the configuration errors in the configs before they bite you... You will ultimately experience more troubleshooting with the use of a diverse POD. Jut my .02 like I said. I think the Proctor lab PODS setup is awesome and thus I use both, but feel that if I had to choose one over the other I would definitely want my own POD. Thanks, Justin McIntyre Message: 2 Date: Mon, 11 Jun 2012 12:16:38 -0500 From: Steve Nicklas steve.nickl...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] using own lab equipment (servers) and troubleshooting exercises Message-ID: CANwcTAXMLvEZtLc-=16hc29om71m99cpvsru52k9f8uh_za...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hello all, When using your own servers, is it still possible to fully experience the troubleshooting sections in the labs? With IOS devices, of course it is easy to load up the intentionally flawed config file to the router to start troubleshooting. But with a CUCM for example, how can this be done? Or is this a concern? Thanks, Steve This email and any files transmitted with it are confidential and are intended for the sole use of the individual to whom they are addressed. Black Box Corporation reserves the right to scan all e-mail traffic for restricted content and to monitor all e-mail in general. If you are not the intended recipient or you have received this email in error, any use, dissemination or forwarding of this email is strictly prohibited. If you have received this email in error, please notify the sender by replying to this email. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] no trace found but calls get routed br1-Hq
Qos doesn't stop calls from completing - it just impacts how packets are prioritized for sending / dropping / etc. CAC and qos work together. If you admit a call on a maxed-out link, all the calls will degrade not just the new call. QOS can help keep non-voice traffic from impacting voice traffic. CAC keeps voice traffic from impacting other voice traffic. In your case, it depends where you press hold, if you're using MOH, and where the MOH source is. If you pressed hold on the branch phone, then the HQ phone connected to music at HQ. There was no media traversing the wan for the first call while held. DQ d...@cisco.com Sent from my iPhone On Jun 11, 2012, at 4:52 PM, Krishna vinayak_...@yahoo.com wrote: Thank you Dan for providing me the detail info. I assumed the same but not sure with my hypothesis. I am wondering if this is the case, then wan qos will not be able to do much isn't it... for instance yesterday i configured llq-cbfwq with bandwidth of 28 for rtp traffic between hq and br1. And, when i called the 5002 from 1002 the call went thru, and this call put on hold and placed another call and it works fine as well. So, from this analysis can i come to conclusion that only location based cac, or rsvp cac can only the number of calls between these two sites??? Thank you Krishna. From: Dan Quinlan (daquinla) daqui...@cisco.com To: Krishna vinayak_...@yahoo.com Cc: ccie_voice@onlinestudylist.com Sent: Monday, June 11, 2012 2:03 PM Subject: Re: [OSL | CCIE_Voice] no trace found but calls get routed br1-Hq You answered your own question. Both DNs are registered to CUCM and are in partitions that the other's CSS can see. The signaling is between each phone and UCM. The media is built directly phone to phone. If CAC failed the call setup, then AAR could be invoked to use the PSTN. Since CAC allows the call, the gateways aren't involved in the call (other than providing IP network connectivity.) DQ d...@cisco.com Sent from my iPhone On Jun 11, 2012, at 2:52 PM, Krishna vinayak_...@yahoo.com wrote: Hi folks, I couldn't understand the call flow between HQ and BR1 which are provisioned/registed in the cucm. here is the detail structure: HQ-phone1 -5002 css-hq-international pt-pt-internal BR1-phone1-1002 css-br1-ld pt-pt-internal Both phones are residing in the partition pt-internal, and br1 is a mgcp site and whereas the hq is the h323 site. when i call 1002 from 5002 or vice versa the call works fine, but when i enable deb isdn q931 or deb voip dialp, i dont see anything. Whereas when i enable RSVP based CAC, i can see the traces with the show sccp connections. could any one help me out how the calls are working in between these two. is it because the phones are registered to cucm, but logically in a different device pool and therefore it routes directly on cucm your help is much appreciated. Thank you. Krishna. ___ 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] [Resolved] UCCX - Do CTI Ports in Partition work ?
Thanks Gurpreet, Dan Krishna. This is now fixed. As Dan mentioned the CSS of caller matters. == When a CTI Route point, redirects the call to the CTI port, the CSS of the device that calls the Route point is used to search for the CTI Port. == From: Pavan K pav.c...@gmail.com Date: Mon, Jun 11, 2012 at 1:33 PM To: ccie_voice@onlinestudylist.com With UCCX, did anybody get calls to work when the CTI ports are in a partition ? If so what CSS did you have to configure ? I have created a RoutePoint in the NULL Partition and CTI ports in a UCCX partition Added a CSS for the RoutePoint that includes the UCCX partition (on both the line device) but the call doesn't connect. If i take the CTI ports out of the partition, everything works perfectly. TIA -- - Pavan -- From: Gurpreet Singh Kukreja tycoononway1...@gmail.com Date: Mon, Jun 11, 2012 at 3:27 PM To: Pavan K pav.c...@gmail.com Cc: ccie_voice@onlinestudylist.com Hi Pavan, We've seen this behavior with UCCX. Logically, the calls should work w/ or w/o partition applied on the CTI Port Group, keeping in mind the CSS applied on the CTI Route Point. Few things to keep in mind: 1) Always apply the changes on these Triggers/ Port Groups from the CCX and never from the CM. 2) If you apply the correct CSS on the Trigger which includes the partition of the Port group, the calls should work. 3) Even after applying the changes if the calls do not work, it could be very possible that the changes you're making from the CCX are not getting updated on the CM. In this case, first run the Data Resync from the CCX and make sure there are no exceptions in the output. Then, restart the CTI Manager on all CM servers and then restart the CCX Engine. - Gurpreet ___ 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 -- From: Dan Quinlan (daquinla) daqui...@cisco.com Date: Mon, Jun 11, 2012 at 3:34 PM To: Pavan K pav.c...@gmail.com Cc: ccie_voice@onlinestudylist.com I would think that the inbound caller (ip phone or gw) would need the CSS to access the CTI ports. DQ d...@cisco.com Sent from my iPhone -- From: Krishna vinayak_...@yahoo.com Date: Mon, Jun 11, 2012 at 3:53 PM To: Pavan K pav.c...@gmail.com pavan, I worked on uccx lab and it worked fine for me. All that you need to remember one point always, what does the CTI Route point has to see. in this case the CTI route point has to see the phones partition in order to handover the calls to the phone agents. Check that internal dns are listed in your css to make this work. thank you krishna. From: Pavan K pav.c...@gmail.com To: ccie_voice@onlinestudylist.com Sent: Monday, June 11, 2012 1:33 PM Subject: [OSL | CCIE_Voice] UCCX - Do CTI Ports in Partition work ? ___ 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 -- - Pavan ___ 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] FileOpen for iPad
FYI - FileOpen is now available for iPad. It works great for your IPExpert files. DQ d...@cisco.com Sent from my iPhone ___ 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] [Resolved] UCCX - Do CTI Ports in Partition work ?
Hi Pavan, I would like to add a few things here. The only place the CSS of the caller matters is when they're calling the CTI Route Point and not the ports. Let me give an example. Consider the below call flow: Caller's Cell Phone Dials the 10 digit # for the Co. PRI MGCP G/W 4 digits enter the CM Matches a Translation Pattern Forwards the Calls to the CTI Route Point of the CCX CTI port answers the call and then the Agent picks up the call. Now think at what all places do we need a CSS and what partition we need to hit: 1) G/W's Inbound CSS should have the PT for the TP (If any) 2) TP's CSS should have the PT of the CTI Route Point. 3) CTI Route Point's CSS should have the PT of the CTI Ports. 4) Finally, CTI Port's CSS should have the Partition of the Agent's Phone. In this scenario, Endpoint's CSS matters to reach the next destination. Hope this makes sense. Regards Gurpreet On Mon, Jun 11, 2012 at 5:33 PM, Pavan K pav.c...@gmail.com wrote: Thanks Gurpreet, Dan Krishna. This is now fixed. As Dan mentioned the CSS of caller matters. == When a CTI Route point, redirects the call to the CTI port, the CSS of the device that calls the Route point is used to search for the CTI Port. == From: Pavan K pav.c...@gmail.com Date: Mon, Jun 11, 2012 at 1:33 PM To: ccie_voice@onlinestudylist.com With UCCX, did anybody get calls to work when the CTI ports are in a partition ? If so what CSS did you have to configure ? I have created a RoutePoint in the NULL Partition and CTI ports in a UCCX partition Added a CSS for the RoutePoint that includes the UCCX partition (on both the line device) but the call doesn't connect. If i take the CTI ports out of the partition, everything works perfectly. TIA -- - Pavan -- From: Gurpreet Singh Kukreja tycoononway1...@gmail.com Date: Mon, Jun 11, 2012 at 3:27 PM To: Pavan K pav.c...@gmail.com Cc: ccie_voice@onlinestudylist.com Hi Pavan, We've seen this behavior with UCCX. Logically, the calls should work w/ or w/o partition applied on the CTI Port Group, keeping in mind the CSS applied on the CTI Route Point. Few things to keep in mind: 1) Always apply the changes on these Triggers/ Port Groups from the CCX and never from the CM. 2) If you apply the correct CSS on the Trigger which includes the partition of the Port group, the calls should work. 3) Even after applying the changes if the calls do not work, it could be very possible that the changes you're making from the CCX are not getting updated on the CM. In this case, first run the Data Resync from the CCX and make sure there are no exceptions in the output. Then, restart the CTI Manager on all CM servers and then restart the CCX Engine. - Gurpreet ___ 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 -- From: Dan Quinlan (daquinla) daqui...@cisco.com Date: Mon, Jun 11, 2012 at 3:34 PM To: Pavan K pav.c...@gmail.com Cc: ccie_voice@onlinestudylist.com I would think that the inbound caller (ip phone or gw) would need the CSS to access the CTI ports. DQ d...@cisco.com Sent from my iPhone -- From: Krishna vinayak_...@yahoo.com Date: Mon, Jun 11, 2012 at 3:53 PM To: Pavan K pav.c...@gmail.com pavan, I worked on uccx lab and it worked fine for me. All that you need to remember one point always, what does the CTI Route point has to see. in this case the CTI route point has to see the phones partition in order to handover the calls to the phone agents. Check that internal dns are listed in your css to make this work. thank you krishna. From: Pavan K pav.c...@gmail.com To: ccie_voice@onlinestudylist.com Sent: Monday, June 11, 2012 1:33 PM Subject: [OSL | CCIE_Voice] UCCX - Do CTI Ports in Partition work ? ___ 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 -- - Pavan ___ 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] callback
Hi Joel, Are there two call manager servers? Did the phones flap between servers (Pub Sub) cos of any service restarts? Are the phone currently registered to there Primary CM server? The Cisco Call Back feature allows you to receive call back notification on your Cisco IP Phone when a called party line becomes available. To receive call back notification, a user presses the CallBack softkey while receiving a busy or ringback tone. You can activate call back notification on a line on a Cisco IP Phone within the same Cisco CallManager cluster as your phone. You cannot activate call back notification if the called party has forwarded all calls to another extension. This explains when you can actually press the key. Finally, make sure Cisco Extended Functions is running on the servers. Also try going through this link which explains some t/shooting steps (although old but should help): http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_qanda_item09186a0080191057.shtml#q17 What CM version are you using? Regards Gurpreet On Sun, Jun 10, 2012 at 6:13 PM, Joel Petralia jrpetra...@msn.com wrote: I am trying to get the callback feature working but the phones keep displaying Callback in not active. Any help would be greatly appreciated. Thank you, Joel ___ 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] PSTN issues
Hi Leslie, Try removing any policy map commands if they exist and make sure the Vlans are in place. You can enter the policy map commands if any later if the phones get registered. Regards Gurpreet On Fri, Jun 8, 2012 at 12:56 AM, Leslie Meade leslie.me...@lvs1.com wrote: I have a strange issue that I think I know the issue but do not know how to fix it. I have a test lab and today I fired up my PSTN router a 3745, and for the past year it was worked with out an issue. But today i am getting the phone hanging at requesting Softkey Template then it will cycle through again There has been no changes to the router. When i do a debug tftp events i get the following Jun 7 21:35:57.695: New Skinny socket accepted [1] (1 active) Jun 7 21:35:57.695: sin_family 2, sin_port 50667, in_addr 10.10.200.21 Jun 7 21:35:57.695: skinny_add_socket 1 10.10.200.21 50667 Jun 7 21:35:57.703: %IPPHONE-6-REG_ALARM: 17: Name=SEP0014F26A78CA Load=8.0(9.0) Last=KeepaliveTO Jun 7 21:35:57.703: ephone-(3)[2] StationRegisterMessage (1/1/5) from 10.10.200.21 Jun 7 21:35:57.703: ephone-(3)[2] Register StationIdentifier DeviceName SEP0014F26A78CA Jun 7 21:35:57.703: ephone-(3)[2] StationIdentifier Instance 1 deviceType 7 Jun 7 21:35:57.703: ephone-3[1]:stationIpAddr 10.10.200.21 Jun 7 21:35:57.703: ephone-3[1][SEP0014F26A78CA]:maxStreams 0 Jun 7 21:35:57.703: ephone-3[1][SEP0014F26A78CA]:From Phone raw protocol Ver 0x856B Jun 7 21:35:57.703: ephone-3[1][SEP0014F26A78CA]:protocol Ver 0x856B Jun 7 21:35:57.703: ephone-3[1][SEP0014F26A78CA]:phone-size 5480 dn-size 688 Jun 7 21:35:57.703: ephone-(3) Allow any Skinny Server IP address 10.10.250.2 Jun 7 21:35:57.703: ephone-3[1][SEP0014F26A78CA]:Found entry 2 for 0014F26A78CA Jun 7 21:35:57.703: ephone-3[1][SEP0014F26A78CA]:socket change 1 to 2 Jun 7 21:35:57.703: ephone-3[1][SEP0014F26A78CA]:DisAssociate: Closed socket 1 while REGISTERED Jun 7 21:35:57.707: %IPPHONE-6-UNREGISTER_ABNORMAL: ephone-3:SEP0014F26A78CA IP:10.10.200.21 Socket:1 DeviceType:Phone has unregistered abnormally. Jun 7 21:35:57.707: ephone-3[-1][SEP0014F26A78CA]:FAILED: CLOSED old socket -1 Jun 7 21:35:57.707: ephone-3[2][SEP0014F26A78CA]:***Force device subtype to 0 Jun 7 21:35:57.707: ephone-3[2][SEP0014F26A78CA]:phone SEP0014F26A78CA re-associate OK on socket [2] Jun 7 21:35:57.707: %IPPHONE-6-REGISTER: ephone-3:SEP0014F26A78CA IP:10.10.200.21 Socket:2 DeviceType:Phone has registered. Jun 7 21:35:57.707: Phone 2 socket 2 Jun 7 21:35:57.707: Skinny Local IP address = 10.10.250.2 on port 2000 Jun 7 21:35:57.707: Skinny Phone IP address = 10.10.200.21 50667 Jun 7 21:35:57.707: ephone-3[2][SEP0014F26A78CA]:Signal protocol ver 8 to phone with ver 11 Jun 7 21:35:57.707: ephone-3[2][SEP0014F26A78CA]:Date Format M/D/Y Jun 7 21:35:57.707: ephone-3[2]:RegisterAck sent to sockettype ephone socket 2: keepalive period 30 use sccp-version 8 Jun 7 21:35:57.707: ephone-3[2]:CapabilitiesReq sent Jun 7 21:35:57.715: ephone-3[2]:MediaPathEventMessage Jun 7 21:35:57.715: ephone-3[2]:MediaPathEventMessage Jun 7 21:35:57.719: ephone-3[2]:MediaPathEventMessage Jun 7 21:35:57.755: ephone-3[2]:MediaPathEventMessage Jun 7 21:35:57.759: ephone-3[2]:MediaPathEventMessage Jun 7 21:35:57.919: ephone-3[2]:CapabilitiesRes received Jun 7 21:35:57.919: ephone-3[2][SEP0014F26A78CA]:Caps list 8 WideBand_256K 120 ms G711Ulaw64k 40 ms G711Alaw64k 40 ms G729AnnexB 60 ms G729AnnexAwAnnexB 60 ms G729 60 ms G729AnnexA 60 ms Unrecognized Media Type 257 4 ms Jun 7 21:35:57.919: ephone-3[2]:MediaPathEventMessage Jun 7 21:35:57.919: ephone-3[2]:MediaPathEventMessage Jun 7 21:35:57.919: ephone-3[2]:ButtonTemplateReqMessage Jun 7 21:35:57.919: ephone-3[2][SEP0014F26A78CA]:StationButtonTemplateReqMessage set max presentation to 6 Jun 7 21:35:57.919: ephone-3[2]:CheckAutoReg Jun 7 21:35:57.919: ephone-3[2]:AutoReg is disabled Jun 7 21:35:57.919: ephone-3[2][SEP0014F26A78CA]:Setting 6 lines 0 speed-dials on phone (max_line 6) Jun 7 21:35:57.919: ephone-3[2][SEP0014F26A78CA]:First Speed Dial Button location is 0 (0) Jun 7 21:35:57.919: ephone-3[2][SEP0014F26A78CA]:Configured 0 speed dial buttons Jun 7 21:35:57.919: ephone-3[2]:ButtonTemplate lines=6 speed=0 buttons=6 offset=0 Jun 7 21:35:57.927: ephone-3[2]:StationSoftKeyTemplateReqMessage Jun 7 21:35:57.927: ephone-3[2]:StationSoftKeyTemplateResMessage -- This is where the Requesting Softkey Template hangs Jun 7 21:35:58.115: Bring up DN 1 by SkinnyCheckDnStatus Jun 7 21:35:58.591: TFTP: Finished system:/its/XMLDefault7960.cnf.xml, time 00:00:25 for process 154 Jun 7 21:35:59.115: Bring up DN 2 by SkinnyCheckDnStatus Jun 7 21:36:00.115: Bring up DN 3 by SkinnyCheckDnStatus Jun 7 21:36:01.115: Bring up DN 4 by SkinnyCheckDnStatus Jun 7 21:36:02.115: Bring up DN 5 by SkinnyCheckDnStatus Jun 7 21:36:02.623: TFTP: Finished
Re: [OSL | CCIE_Voice] FileOpen for iPad
Hey Dan, Have you tested? I tested Fileopen few weeks ago, but i couldn't open the IPE files. On Mon, Jun 11, 2012 at 7:56 PM, Dan Quinlan (daquinla) daqui...@cisco.comwrote: FYI - FileOpen is now available for iPad. It works great for your IPExpert files. DQ d...@cisco.com Sent from my iPhone ___ 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 -- Rafael Chavantes ___ 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] DTMF from BR1 phone to CUC via SIP trunk not recognized properly
Hey Guys, When I call CUC pilot from BR1 phone, the dtmf tones are not recognized properly by CUC, i.e. BR1 phone cannot login to mailbox or select any other option via DTMF. If I remove crtp, everything works fine. Topology: SCCP phone(BR1 site) à g729r8 with crtp à CUCM à SIP trunk(with OOB and RFC2833 as dtmf options) à CUC Things I have tried so far, 1) All dtmf options in SIP trunk. 2) Enabled mtp option 3) In CUC, changed codec type to just g711u, just g729 and both(which is the default). I found other posts on this issue but none of them has the solution. Thanks, Tapan ___ 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] no trace found but calls get routed br1-Hq
Well that is true unless you drop the packets while doing QOS : ) From: Dan Quinlan (daquinla) daqui...@cisco.com Date: Monday, June 11, 2012 4:17 PM To: Krishna vinayak_...@yahoo.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] no trace found but calls get routed br1-Hq Qos doesn't stop calls from completing - it just impacts how packets are prioritized for sending / dropping / etc. CAC and qos work together. If you admit a call on a maxed-out link, all the calls will degrade not just the new call. QOS can help keep non-voice traffic from impacting voice traffic. CAC keeps voice traffic from impacting other voice traffic. In your case, it depends where you press hold, if you're using MOH, and where the MOH source is. If you pressed hold on the branch phone, then the HQ phone connected to music at HQ. There was no media traversing the wan for the first call while held. DQ d...@cisco.com Sent from my iPhone On Jun 11, 2012, at 4:52 PM, Krishna vinayak_...@yahoo.com wrote: Thank you Dan for providing me the detail info. I assumed the same but not sure with my hypothesis. I am wondering if this is the case, then wan qos will not be able to do much isn't it... for instance yesterday i configured llq-cbfwq with bandwidth of 28 for rtp traffic between hq and br1. And, when i called the 5002 from 1002 the call went thru, and this call put on hold and placed another call and it works fine as well. So, from this analysis can i come to conclusion that only location based cac, or rsvp cac can only the number of calls between these two sites??? Thank you Krishna. From: Dan Quinlan (daquinla) daqui...@cisco.com To: Krishna vinayak_...@yahoo.com Cc: ccie_voice@onlinestudylist.com Sent: Monday, June 11, 2012 2:03 PM Subject: Re: [OSL | CCIE_Voice] no trace found but calls get routed br1-Hq You answered your own question. Both DNs are registered to CUCM and are in partitions that the other's CSS can see. The signaling is between each phone and UCM. The media is built directly phone to phone. If CAC failed the call setup, then AAR could be invoked to use the PSTN. Since CAC allows the call, the gateways aren't involved in the call (other than providing IP network connectivity.) DQ d...@cisco.com Sent from my iPhone On Jun 11, 2012, at 2:52 PM, Krishna vinayak_...@yahoo.com wrote: Hi folks, I couldn't understand the call flow between HQ and BR1 which are provisioned/registed in the cucm. here is the detail structure: HQ-phone1 -5002 css-hq-international pt-pt-internal BR1-phone1-1002 css-br1-ld pt-pt-internal Both phones are residing in the partition pt-internal, and br1 is a mgcp site and whereas the hq is the h323 site. when i call 1002 from 5002 or vice versa the call works fine, but when i enable deb isdn q931 or deb voip dialp, i dont see anything. Whereas when i enable RSVP based CAC, i can see the traces with the show sccp connections. could any one help me out how the calls are working in between these two. is it because the phones are registered to cucm, but logically in a different device pool and therefore it routes directly on cucm your help is much appreciated. Thank you. Krishna. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com http://www.PlatinumPlacement.com ___ 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
