Re: [OSL | CCIE_Voice] SIP Gateway with Unity Connection issues
So what does the diversion header get translated to when you try the call via UCxN? Are you saying that the SIP profile is working when you directly call the forwarded phone but not when UCxN AA calls the forwarded phone. Can we see the comparing SIP traffic. Can we see associated SIP traffic when you call forwarded phone and then the SIP traffic when UCxN AA makes the call? I'd like to see the difference. Based upon your diversion header info being set to .*@.* this should apply the change to the UCxN VM pilot as well depending on the length of your VM pilot etc... If your VM pilot is only 4 digits and not 7 then that may be the reason it works by calling forwarded phone directly but not UCxN AA. Maybe Provider isn't seeing enough incoming digits? Excluding all of these options you could also check and see if your provider allows additional authentication methods for calls. Trunk groups, digest authentications etc... Thanks, Justin 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
[OSL | CCIE_Voice] BACD Timer
http://blog.ipexpert.com/2009/01/24/b-acd-in-a-nutshell/ check this out. Thanks, Justin 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] CCIE_Voice Digest, Vol 88, Issue 91
Pressing the ? Twice will in fact give u the results your looking for. The region setting on UCM will set the codec that u suoport during call setup on the phone and the system param settings for UCCX set the codec you support for call setup. Thus as long as you have overlapping or at least one of the same codecs supported on both sides of connection the call will be able to setup the rtp connection between each. Pressing ? twice onxe rtp session is established gives you the codec that was negotiated. Keep in mond that the codec will change as different prompts or MOH are transgressed between the UCM and UCCX server. Thanks, Justin McIntyre 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
[OSL | CCIE_Voice] CUCM GK Port Number
With regards to your service parameter. Just make sure that the name of your trunk is listed in this service parameter. Then re-register your UCMs to the Gatekeeper. Couldn't also hurt to shut / no shut the gatekeeper. The service parameter is the key to what you are looking for here. Depending on what time frames you have entered for your registration timeouts you should see it repair after some time as long as you have the service parameter configured correctly. Thanks, Justin 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
[OSL | CCIE_Voice] Phone not pick up TFTP or config from
I seem to remember hitting an issue like this in the past before. In my case, I did a factory reset. Hold # during power on, 123456789*0#and wait. Just make sure that prior to doing this, you have troubleshot everything on the network side first. Additionally, make sure you do have the dhcp and option 150 working correctly. I've heard horror stories of doing this to a phone and it never coming back from the black screen so do some research on the subject first. Again, in my case, phone went black for about 20 minutes and eventually came back up, pulled down its config and registered. This is a quick stab at a fix but if it were me, I would pull some traces first. When and if you take your lab attempt, you will need to be comfortable with the various troubleshooting procedures anyhow. Couple things to look at. 1.) Launch the RTMT and take a look at the TFTP counters. Verify or clear the counters and then reset the phone. Does the counter increment? If it does can you guarantee that it is this phone that is pulling the TFTP config down. a. If you can verify TFTP is working, you may still have a DB replication issue. Change the UCM group in the DP/phone to point at PUB first. Reset phone and see if it registers. If it does, there's a chance (and with this version of UCM a good one!!) that your DB is out of sync. (utils db replication stop on sub, stop on pub and then Utils dbreplication force data sync sub all, Vik has a really good write-up on this, search it out). 2.) try to delete the phone and make it a SIP phone. This will make it switch firmware which may get it past the point of where it is getting stuck in the registration process (assuming you don't have a DB issue). Then switch it back to SCCP and go at it again. (however, I would try to verify if TFTP is working or not first) 3.) There are also some SDI/SDL traces that you can pull to check the registration process. (Personally I like to use wireshark for this) 4. ) if you want to use wire shark, there's a couple ways you can do it. a. Span the port connected to your phone to the port connected to your pc and monitor the traffic with wireshark (this depends on the switch that you are using and obviously may vary in method If you are using proctor labs. If you're using proctor labs, ssh into the OS of the tftp server and launch a file capture and reset phone. Use RTMT to download the capture from the server and open with wireshark). Do this on the subscriber too to check registration process). Note: spanning ports and vlans on a Cisco switch can be seriously detrimental to the I/O of the device. Research it out a bit. b. Turn on port mirroring in the phone. If the phone is in fact taking the tftp config but just not registering with UCM, you may be able to turn on Port mirroring/spanning in the phone config page and then connect your pc to the PC port of the phone and turn on wire shark. In my opinion, all of these are subjects and techniques that a CCIE should be comfortable with, so don't be afraid to delve a little outside the workbooks to practice something. Ultimately, this will make you more aware of exactly what is happening with the phones/system rather than just figuring out how to complete a particular section of a workbook. Thanks, Justin 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] CUCM GK Port Number
My take based upon the written requirements is that the screen shot is erroneous and 1720 was intended. Thanks, Justin McIntyre CCIE #36706 Engineer Mutual Telecom Services Inc. A wholly owned subsidiary of Black Box justin.mcint...@blackbox.commailto:justin.mcint...@blackbox.com COMM: 434-946-1562 DSN: 312-237-1562 Cell: 434-381-0024 On Jun 13, 2013, at 8:02 PM, Martin Sloan martinsloa...@gmail.commailto:martinsloa...@gmail.com wrote: Thanks, Bill. My problem with this one is that the trunks aren't registered under port 1720. They're ports 40446 and 35246. I read 'Port number matching is required' to mean my trunks must match those in the output in the task. With those port numbers I don't believe it's possible. Do you agree it's probably an error in the output on this one? I think the trunks are supposed to be registered under port 1720, but I hate to assume. I know (okay, I've heard) the exam tasks leave some room for interpretation but I think on this one, there's only one way to interpret. Thanks, Marty On Thu, Jun 13, 2013 at 6:52 PM, Bill Lake whl...@gmail.commailto:whl...@gmail.com wrote: Marty, You got to the first column and select System parameters then scroll Dow the the H323 section and there will be your 1720 and there you will put the name of the gk trunk setup in your trunk config Reset the trunk and it should start using 1720 Sent from my iPhone On Jun 13, 2013, at 4:11 PM, Martin Sloan martinsloa...@gmail.commailto:martinsloa...@gmail.com wrote: Justin, Thanks for the assist! I'm still lost on the requirement for this task b/c based on the output supplied, the CUCM's register with GK using ports 40446 and 35246 (gk-trunk_1 and gk-trunk_2). I'm not sure if: 1) I'm misunderstanding the question and/or output 2) The output in the task was supposed to show the CUCM's registered with port 1720 3) There is really a way to register the CUCM's with the ports they have in the output and I don't know how to do it. I've attached a SS of the section. Can you make any suggestions? I'm leaning heavily towards option 1 :-\ Thanks, Marty On Thu, Jun 13, 2013 at 4:04 PM, Justin McIntyre justin.mcint...@blackbox.commailto:justin.mcint...@blackbox.com wrote: With regards to your service parameter. Just make sure that the name of your trunk is listed in this service parameter. Then re-register your UCMs to the Gatekeeper. Couldn't also hurt to shut / no shut the gatekeeper. The service parameter is the key to what you are looking for here. Depending on what time frames you have entered for your registration timeouts you should see it repair after some time as long as you have the service parameter configured correctly. Thanks, Justin 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.comhttp://www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.PlatinumPlacement.com Vol2_Lab6_Task4.2.png ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] *** MTP Understanding. *** quote To MTP or
Think about it this way. When you place hold you are telling the opposite phone/gateway to listen to a new RTP stream. So where is that new RTP stream being sourced from. If it's coming from your PUB/SUB then what device pool is that MOH server in and what MRGL/group is within that Device Pool? If it is structured such that it would be definitive to use the MTP then yes you would also be terminating that RTP stream to the MTP as well. Just a side note , which I'm sure you're already aware of, the PUB and SUB built in software MTPs only support g711. Hope this helps. Thanks, Justin McIntyre 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] CME Multicast MOH Port number
So it is not a typo. This may clear things up a bit. Let's play out two scenarios 1st Scenario Standalone CME site. Router is configured with CME and has 2 ephones registered. Your customer has complained that having the router source an individual UNICAST steam each time someone places someone else on hold is just beyond what they want to do. Instead they have asked to you multicast the MOH so that there is one stream all the time that the end client may tune into whilst placed on hold. To do this you go into the CME and enter the following commands... Telephony-service Moh music-on-hold.au Multicast moh 229.0.0.1 port 2000 route x.x.x.x x.x.x.x ... At this point the router is now streaming multicast traffic. You must also enable PIM on the loopback such that the MOH can be Multicast to any calls across the PSTN. But for the most part , there is now active multicast MOH steam on your router. Now when an individual places another person on hold the traffic flow goes as such Phone1 presses hold-sccp/hold signal sent to CME for active call-CME Router CME router looks through parameters sees that Multicast MOH is enabled and tells distant end to listen to 229.0.0.1 and port 2000 for audio. Basically it tells the phone to listed on a new port and IP such that it can hear the rtp of the MOH. Phone1 presses hold again-sccp/hold signal sent to CME (give me my call back...)CME Router CME router then tells distant phone, ok your done listening to that IP and port, go back to the original port and IP again. But in the back ground the MOH is still being multicasted, the distant phone just isn't listening anymore. The commands entered into CME earlier do two things.. 1. Tells the router to stream MOH as multicast and 2. Gives a Multicast IP address and Port number to multicast on as well as inform the placed on hold caller to which IP and port to listen to. Now Scenario two. You have a Cisco UCM running in a Centralized location with distributed call processing. IE.. you have users at a remote location that home back to the HQ location in order to register and process call requests. Those remote users also have a local gateway which is also providing SRST services. Router with CME/CMFB running SRST. Now let say that your customer says we have noticed that when a large number of Unicast streams of MOH are sent across our limited WAN that it is detrimental to our link speeds. However, we don't want to have multicast traffic going across either because it is not a valid solution for us. In this case you can trick the users at the remote site into thinking that they are listening to MOH from the HQ location although it is actually being sourced from the local gateway. So lets think about how this happens. Usually you would set either your PUB or your SUB at the HQ location to Multicast an MOH file. You would then sent th e number of HOPS to reach from one subnet/VLAN across your WAN link and into your voice VLAN on the remote location. You would also setup PIM on the appropriate interfaces to allow the Multicast traffic to traverse the Layer 3 interfaces. Then the call flow would happen as such for an active call Phone 1 send hold signal to UCM cluster at HQ location across WAN-UCM UCM then checks device pool settings of user/gateway that is being placed on hold to find that they should listen to Multicast MOH UCM sends signal to distant phone/gateway to tune into 229.0.0.1 port 16384--phone/gateway. The phone/gateway then hears the Multicast traffic that is already actively being transmitted(this is why you hear the music on hold in the middle of the song rather than it starting from beginning). Now we're almost there, no unicast across the WAN, however we're still breaking our customer's requirement of no Multicast either. Ok so if I turn off PIM then I will stop the Multicast but the user place on hold with no longer hear the stream... So repeat the same steps for scenario one. However there are a few things to consider here. The users at the remote site are registered to the UCM at HQ location. This means all signaling for call activity is based upon how the UCM cluster is configured. So when you setup your Multicast MOH in your CME/SRST router you must match the Multicast settings that you set up back on the SUB/PUB upon which you are/were multicasting your MOH file from. Key details here are codec, Multicast IP, and port number. The Multicast IP and port you set in CME/SRST needs to match what the UCM thinks that distant user is using. So if you have increment on IP address or Port number selected results may vary. Research this if you are confused here. Ok so now your CME router is now playing Multicast MOH from flash on the same IP and port as what UCM was sending across the WAN. You also stopped the multicast traffic from playing across the WAN.
Re: [OSL | CCIE_Voice] SRST and ephone-template problem
Are you applying the template once the phones have already registered and then trying to reset them? Are you resetting them at all while they are in SRST mode? 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
[OSL | CCIE_Voice] Passed part 2
and hone in my techniques even finer. I would study a few days a week up to my second lab attempt, but not for more than about 10 to 12 hours a week. However in the final week I practiced about 2 , 8 hour mock labs, just to make sure i was still in my targeted time frames. The last two days before the test i did nothing. In between each lab attempt i took about a 6 week period. The night before the test i took my wife out to a local Brazilian steak house i had found the previous time and then cut back to the hotel for an early nights rest. Fat chance of that happening. I t ossed and turned for a good part of the night... am i truly ready, what if i run out of time again, should i have waited longer, etc... . but i finally put those thoughts to rest and slept for a good little while. The following morning I awoke with the realization that I would either pass this time or i wouldn't , greater people than I had taken many more attempts to pass, why was I any better? So i went in this time with a renewed spirit. I was very at ease and did not let my self get too worked up. So there i sat at 7 am in RTP with the rest of the folks, waiting on the proctor. Everyone was focused and nervous. So I did what i thought would help and started talking to everyone. Where are you from, what test are you taking etc... and so on. In all actuality i was trying to lighten the mood for everyone in the room including my self. It's hard to be too nervous when you have my Alabama accent talking about technology. So then in walked the proctor... here's th e bathroom, here's the food, run that way if there's a fire, ready set go! So i sat down at my station opened a terminal window and ABSOLUTELY FROZE UP Oh crap, why am i not typing?? Type something Justin, anything Then I said a prayer, I gave in to let go and let God, had a moment and began to go to work. I knew that being at ease would be the difference for me this time so i refused to look at the clock at all. I was moving along pretty good and felt pretty good about my chances this time. I would get to a new task and silently go yes, i know this one...i think. It's probably best to be respectful of the test at all times or she'll catch you when your not paying attention and pants you in front of all your buddies. I'm just saying... So lunch came , we all sat down, gave out some jokes, talked about traveling, and then went back to work again. This time I actually finished about 1.5 hours before time was up. I then went back and tried to fail all o f my questions that I had completed. I tried to pick apart everything I had completed as if I truly wanted to mark the question wrong. This helped tremendously, i found several small mistakes that would have meant a big fat goose egg for that question. I left that day with a smile on my face. I knew it was in God's hands, but I knew i had given everything i could to that keyboard at that station. So completely spent for the day i laid down with hopes of a good nights restWRONG!!! I tossed and turned. So i checked my email every 5 minutes until my wife threatened to throw my IPAD out the window. I slept some but finally just got up at about 5 am and watched a little TV. So as i'm sure you don't want to keep hearing my complete life story... I received an email the next morning from Cisco...please click here for your results. I clicked and i conquered So that's it. I spent the rest of that day telling everyone that i knew about my accomplishment , I n ever could find one of those planes that writes stuff in the sky, but i think i got just about everyone else covered. P.S. I slept like a baby that night!! Thanks, Justin McIntyre CCIE Voice #36706 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] CCIE_Voice Digest, Vol 79, Issue 87
I did the video in my evenings and audio when driving to and from work. Thanks, Justin McIntyre Engineer Mutual Telecom Services Inc. A wholly owned subsidiary of Black Box justin.mcint...@blackbox.com COMM: 434-946-1562 DSN: 312-237-1562 Cell: 434-381-0024 On Sep 23, 2012, at 9:26 PM, ccie_voice-requ...@onlinestudylist.com 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. Re: Passed part 2 (Justin McIntyre) (Kamran Ahsanullah) -- Message: 1 Date: Sun, 23 Sep 2012 22:04:00 +0300 From: Kamran Ahsanullah kamran.ahsanul...@gmail.com To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Passed part 2 (Justin McIntyre) Message-ID: camxg9ifgtycmcvdxzn-0wqggwgmxakydtyxqshzqxnjjbey...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 thanks for the detailed write up Justin, particularly in your study approach. How did u handle the audio and video on demand, did u do the audio or video's 1st? enjoy getting back to your wife and fishing! On 23 September 2012 18:32, 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. WB 2 Lab 2 UCCX questions (Randall Crumm) 2. Re: CCIE Voice V4 (madhav bhardwaj) 3. Passed part 2 (Justin McIntyre) 4. Re: WB 2 Lab 2 UCCX questions (William Bell) -- Message: 1 Date: Sat, 22 Sep 2012 16:47:07 -0700 (PDT) From: Randall Crumm rrcr...@yahoo.com To: Online Study ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] WB 2 Lab 2 UCCX questions Message-ID: 1348357627.78885.yahoomail...@web124901.mail.ne1.yahoo.com Content-Type: text/plain; charset=iso-8859-1 Hello, 1. I am trying to get ?music-on-hold to work when the caller is in queue. I am able to make the ringback work. Any suggestions? I looked at the DSG and it does not seem to address the MOH in queue. 2. It also says to play the in queue prompt every 15 seconds, and if the call is in queue over 60 seconds?terminate?it. I think this means play prompt 4 times and them terminiate call or does it mean play prompt 3 times and terminiate at 4th delay. 3. I do not see where I can download the WB 2 .aef scripts. ?Can someone help me with this? ? Have a great day! Thanks, Randall -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20120922/c25b4479/attachment-0001.html -- Message: 2 Date: Sun, 23 Sep 2012 07:50:00 +0530 From: madhav bhardwaj ashumad...@gmail.com To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE Voice V4 Message-ID: cajsnxwb6aq9gnuthr9d2x3_eygvb0gw2gdxl0mb7kk3kgkw...@mail.gmail.com Content-Type: text/plain; charset=ISO-8859-1 Vik any commnet from your side ? On 9/22/12, madhav bhardwaj ashumad...@gmail.com wrote: Hi Guys Could you please let me know when Cisco is planning to change CCIE voice version 4 and if they are moving for V4 then what changes they can implement in lab. Thanks Ashu -- Message: 3 Date: Sun, 23 Sep 2012 03:01:37 -0400 From: Justin McIntyre justin.mcint...@blackbox.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Passed part 2 Message-ID: 90543778-70f9-47d1-83db-a3700d95c...@blackbox.com Content-Type: text/plain; charset=us-ascii I just wanted to say thank you to everyone that has sent me best wishes and congratulations. I had a few that asked me to share my story and approach , so since i'm on a 8.5 hour flight I figured now was as good a time as any. November 2011: Passed CCIE written. I spent the remainder of November researching vendors for IE training and allocating materials to complete my at home lab. Thank you IPexpert and EBAY. I
Re: [OSL | CCIE_Voice] passed Monday -Part 1
Yes congrats Kevin! Thanks, Justin McIntyre 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
[OSL | CCIE_Voice] . Passed!!!!
I just wanted to say thanks to everyone that helped me throughout this journey. It has been a very humbling and all inspiring experience to say the least. I received my results during breakfast and have been on cloud 9 ever since Thanks, Justin McIntyre 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] Trancoding
Do you have a voice-class codec applied to your in bound dial-peer? This will keep the transcoder from being invoked. Thanks, Justin McIntyre On Sep 19, 2012, at 6:45 AM, ccie_voice-requ...@onlinestudylist.com 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. Re: Trancoding (Bill Lake) 2. Re: Trancoding (Chrysostomos Christofi) -- Message: 1 Date: Wed, 19 Sep 2012 05:28:30 -0500 From: Bill Lake whl...@gmail.com To: Chrysostomos Christofi ch.christ...@logicom.net Cc: Online Study \(ccie_voice@onlinestudylist.com\) ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Trancoding Message-ID: CADpb93M_qAh6Kg2840VRctLMxyWa9drqG8BHk2iOkgUfm4=1...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-7 Can you give us more information, you say VM does not work but does it work for local callers? does it work for PSTN callers? Or do you not have PSTN callers? Can you dial the VM directly? What are the results of show sccp conn when you do this? We just don't have any information but a partial config and it doesn't work. Not enough to go on. Now this is just a wild guess but I seem to remember someone else having an issue like this and they resolved it by using the loopback and not a sub interface. Of course I could be completely wrong as this is just something that is in my foggy memory but don't think it could hurt to try. On another note, why do you have interface GigabitEthernet0/0.100 encapsulation dot1Q 100 ip address 10.4.2.1 255.255.255.0 h323-gateway voip interface h323-gateway voip h323-id cme h323-gateway voip bind srcaddr 10.4.2.1 you appear to be setting up for a Gatekeeper but do not have the rest of the commands. If this is just an H323 gateway you do not need h323-gateway voip interface or h323-gateway voip h323-id cme On Wed, Sep 19, 2012 at 2:45 AM, Chrysostomos Christofi ch.christ...@logicom.net wrote: Hi ** ** Any update? ** ** Regards ** ** ** ** *From:* Chrysostomos Christofi *Sent:* ?, 18 ??? 2012 1:23 ?? *To:* 'ccie_voice-boun...@onlinestudylist.com' *Subject:* Trancoding ** ** Hi to all ** ** I need your input pls for the below ** ** I have one cluster cucm and ore remote branch ** ** Calls to remode branch are via h2323 with codec g729 Everything its ok except when the call going to voice mail Voice mail is configured with g711ulaw ** ** I have created a transcoding in cme but nothing change What is your opinion ** ** CME CONFIG ! interface GigabitEthernet0/0.100 encapsulation dot1Q 100 ip address 10.4.2.1 255.255.255.0 h323-gateway voip interface h323-gateway voip h323-id cme h323-gateway voip bind srcaddr 10.4.2.1 ! sccp local GigabitEthernet0/0.100 sccp ccm 10.4.2.1 identifier 1 version 7.0 sccp ! sccp ccm group 1 bind interface GigabitEthernet0/0.100 associate ccm 1 priority 1 associate profile 2 register transcode ! dspfarm profile 2 transcode codec g729r8 codec g729br8 codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 maximum sessions 8 associate application SCCP ! telephony-service sdspfarm units 1 sdspfarm transcode sessions 4 sdspfarm tag 2 transcode ** ** CCME#SHow dspfarm all Dspfarm Profile Configuration ** ** Profile ID = 2, Service = TRANSCODING, Resource ID = 1 Profile Description : Profile Service Mode : Non Secure Profile Admin State : UP Profile Operation State : ACTIVE Application : SCCP Status : ASSOCIATED Resource Provider : FLEX_DSPRM Status : UP Number of Resource Configured : 8 Number of Resource Available : 8 Codec Configuration: num_of_codecs:6 Codec : g729r8, Maximum Packetization Period : 60 Codec : g729br8, Maximum Packetization Period : 60 Codec : g711ulaw, Maximum Packetization Period : 30 Codec : g711alaw, Maximum Packetization Period : 30 Codec : g729ar8, Maximum Packetization Period : 60 Codec : g729abr8, Maximum Packetization Period : 60 Dspfarm Profile Configuration ** ** ** ** ** ** ** ** Regards
Re: [OSL | CCIE_Voice] Trancoding
Try taking the Voice Class codec out of the inbound dial-peer that you have here. You should see then then that your transcoder will start to be invoked. I believe what is happening to you is that you are sending a request over to use 729, and then the internal call to CUE is requesting 711u. The transcoder will only be invoked if the inbound dial peer does not support the list of codecs. In your case, the inbound dial-peer does support both codecs (voice-class codec 1), so the transcoder is not being invoked. Simply remove the command voice-class codec1 from your dial-peer listed below and see what happens. Thanks, Justin McIntyre On Sep 19, 2012, at 8:19 AM, Chrysostomos Christofi ch.christ...@logicom.net wrote: Hi I have ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g711alaw codec preference 3 g729r8 ! dial-peer voice 100 voip preference 10 destination-pattern 1... session target ipv4:CUCM IP incoming called-number . voice-class codec 1 dtmf-relay h245-alphanumeric no vad ! But the transcoding doesn't being involved Regards -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Justin McIntyre Sent: Τετάρτη, 19 Σεπτεμβρίου 2012 2:44 μμ To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Trancoding Do you have a voice-class codec applied to your in bound dial-peer? This will keep the transcoder from being invoked. Thanks, Justin McIntyre On Sep 19, 2012, at 6:45 AM, ccie_voice-requ...@onlinestudylist.com 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. Re: Trancoding (Bill Lake) 2. Re: Trancoding (Chrysostomos Christofi) -- Message: 1 Date: Wed, 19 Sep 2012 05:28:30 -0500 From: Bill Lake whl...@gmail.com To: Chrysostomos Christofi ch.christ...@logicom.net Cc: Online Study \(ccie_voice@onlinestudylist.com\) ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Trancoding Message-ID: CADpb93M_qAh6Kg2840VRctLMxyWa9drqG8BHk2iOkgUfm4=1...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-7 Can you give us more information, you say VM does not work but does it work for local callers? does it work for PSTN callers? Or do you not have PSTN callers? Can you dial the VM directly? What are the results of show sccp conn when you do this? We just don't have any information but a partial config and it doesn't work. Not enough to go on. Now this is just a wild guess but I seem to remember someone else having an issue like this and they resolved it by using the loopback and not a sub interface. Of course I could be completely wrong as this is just something that is in my foggy memory but don't think it could hurt to try. On another note, why do you have interface GigabitEthernet0/0.100 encapsulation dot1Q 100 ip address 10.4.2.1 255.255.255.0 h323-gateway voip interface h323-gateway voip h323-id cme h323-gateway voip bind srcaddr 10.4.2.1 you appear to be setting up for a Gatekeeper but do not have the rest of the commands. If this is just an H323 gateway you do not need h323-gateway voip interface or h323-gateway voip h323-id cme On Wed, Sep 19, 2012 at 2:45 AM, Chrysostomos Christofi ch.christ...@logicom.net wrote: Hi ** ** Any update? ** ** Regards ** ** ** ** *From:* Chrysostomos Christofi *Sent:* ?, 18 ??? 2012 1:23 ?? *To:* 'ccie_voice-boun...@onlinestudylist.com' *Subject:* Trancoding ** ** Hi to all ** ** I need your input pls for the below ** ** I have one cluster cucm and ore remote branch ** ** Calls to remode branch are via h2323 with codec g729 Everything its ok except when the call going to voice mail Voice mail is configured with g711ulaw ** ** I have created a transcoding in cme but nothing change What is your opinion ** ** CME CONFIG ! interface GigabitEthernet0/0.100 encapsulation dot1Q 100 ip address 10.4.2.1 255.255.255.0 h323-gateway voip interface h323-gateway voip h323-id cme h323-gateway voip bind srcaddr 10.4.2.1 ! sccp local GigabitEthernet0/0.100 sccp ccm 10.4.2.1 identifier 1 version 7.0 sccp ! sccp ccm group 1 bind interface GigabitEthernet0/0.100
Re: [OSL | CCIE_Voice] Trancoding
Hmm. Lets see. Calls from CUCM to CME phones work fine because the CME phones support both codecs. However , CUE only supports g711ulaw. If you dont mind i'd like to see a couple show commands. Place a call from the CUCM to CME/CUE but run a debug voip dialpeer det on the CME and send us that output. I'd also like to see the dial-peer that you have configured for the CUE. Lets go from there and see what we can do. Thanks, Justin McIntyre On Sep 19, 2012, at 8:42 AM, Chrysostomos Christofi ch.christ...@logicom.net wrote: Hi Justin I have removed the class from the inbound dial peers and the results are the same The transcoding does not get involved Any ideas? -Original Message- From: Justin McIntyre [mailto:justin.mcint...@blackbox.com] Sent: Τετάρτη, 19 Σεπτεμβρίου 2012 3:25 μμ To: Chrysostomos Christofi Cc: ccie_voice@onlinestudylist.com Subject: Re: Trancoding Try taking the Voice Class codec out of the inbound dial-peer that you have here. You should see then then that your transcoder will start to be invoked. I believe what is happening to you is that you are sending a request over to use 729, and then the internal call to CUE is requesting 711u. The transcoder will only be invoked if the inbound dial peer does not support the list of codecs. In your case, the inbound dial-peer does support both codecs (voice-class codec 1), so the transcoder is not being invoked. Simply remove the command voice-class codec1 from your dial-peer listed below and see what happens. Thanks, Justin McIntyre On Sep 19, 2012, at 8:19 AM, Chrysostomos Christofi ch.christ...@logicom.net wrote: Hi I have ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g711alaw codec preference 3 g729r8 ! dial-peer voice 100 voip preference 10 destination-pattern 1... session target ipv4:CUCM IP incoming called-number . voice-class codec 1 dtmf-relay h245-alphanumeric no vad ! But the transcoding doesn't being involved Regards -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Justin McIntyre Sent: Τετάρτη, 19 Σεπτεμβρίου 2012 2:44 μμ To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Trancoding Do you have a voice-class codec applied to your in bound dial-peer? This will keep the transcoder from being invoked. Thanks, Justin McIntyre On Sep 19, 2012, at 6:45 AM, ccie_voice-requ...@onlinestudylist.com 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. Re: Trancoding (Bill Lake) 2. Re: Trancoding (Chrysostomos Christofi) - - Message: 1 Date: Wed, 19 Sep 2012 05:28:30 -0500 From: Bill Lake whl...@gmail.com To: Chrysostomos Christofi ch.christ...@logicom.net Cc: Online Study \(ccie_voice@onlinestudylist.com\) ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Trancoding Message-ID: CADpb93M_qAh6Kg2840VRctLMxyWa9drqG8BHk2iOkgUfm4=1...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-7 Can you give us more information, you say VM does not work but does it work for local callers? does it work for PSTN callers? Or do you not have PSTN callers? Can you dial the VM directly? What are the results of show sccp conn when you do this? We just don't have any information but a partial config and it doesn't work. Not enough to go on. Now this is just a wild guess but I seem to remember someone else having an issue like this and they resolved it by using the loopback and not a sub interface. Of course I could be completely wrong as this is just something that is in my foggy memory but don't think it could hurt to try. On another note, why do you have interface GigabitEthernet0/0.100 encapsulation dot1Q 100 ip address 10.4.2.1 255.255.255.0 h323-gateway voip interface h323-gateway voip h323-id cme h323-gateway voip bind srcaddr 10.4.2.1 you appear to be setting up for a Gatekeeper but do not have the rest of the commands. If this is just an H323 gateway you do not need h323-gateway voip interface or h323-gateway voip h323-id cme On Wed, Sep 19, 2012 at 2:45 AM, Chrysostomos Christofi ch.christ...@logicom.net wrote: Hi ** ** Any update? ** ** Regards ** ** ** ** *From:* Chrysostomos Christofi *Sent:* ?, 18 ??? 2012 1:23
Re: [OSL | CCIE_Voice] Switch QOS query
So I believe your on the right track with your QOS config but there are a few things that need to be modified. 1. I see an issue with your requirements. Have the priority-queue enabled but then also give queue 1 30% bandwidth. If priority-queue out is enabled then this over-rides the bandwidth command for that queue. I know you had some other questions as well specifically about how to drop certain traffic if a queue were 80% full. My suggestion to you would be to review Vik Mahlis QOS blog on the IPEXPERT website. Go to blog.ipexpert.com and select the voice blog on the left. Then look for the QOS section. I think this will clear up most of your questions and get you on your way. Thanks, Justin McIntyre 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] ntp master- is it necessary
I believe you are correct Krishna. My thoughts on it are this. I think the NTP MASTER command is needed else it would not be part of the CLI. However, we must truly understand what the command does before we can decide if it is to be used or not. If a router is configured to sync to an external time source then if NTP master is used the stratum that is configured must be accounted for. At some point down the chain of NTP syncs the stratum will eventually reach a number high enough that the device on the end of the chain will prefer its internal clock as the stratum is better. So if the router is not synchronized to an external source but must act as a NTP source then this is when we want to use the NTP master command. Every situation/requirement is different and we need to ultimately understand the NTP Master command as to whether or not we need to use the command or not. Thanks, Justin McIntyre 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] [OSL|CCIE_Voice] Preferences for hunting past failed PSTN on H.323 GW
So ultimately I believe your assumption is correct. If you were routing from one gateway to another and you receive a legitimate un-allocated number response , then under most circumstances you would absolutely want to stop routing, after all, the number is un-allocated why badger it to death from one angle or another. However, if it were that the result of an un-available route was due to a circuit out of order then by all means we would want to find an alternate path. Reference the Ipexpert BootCamp where I recently learned the depth of this information. For those of you who are wondering the ILT bootcamp and the OWLE that IPEXPERT has to offer is absolutely the best money you could spend. Vik is absolutely in-tune with the ebb and flow of the CCIE Voice and offers priceless insight into the technology. If you want to get a very in-depth look at the information within the Voice blueprint then I suggest you book a class if not both classes as fast as you can. You will come out of them a better technician instead of just someone that can learn a question. Thanks, Justin McIntyre 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
[OSL | CCIE_Voice] [OSL|CCIE_Voice] Passed Dan Quinlan
Congratualtions Sir! I can't imagine how relieved you feel. 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] [OSL|CCIE Voice] no notification sound on ip phone for messages
this means you do not have all configuration completed. You need to check these few places: 1. user licensed for CUP in UCM 2. You have created the Application user for IPPM(PhoneMessenger) in UCM and the phone you are using the IPPM service on is associated with this Application user.. Also make sure this user is CTI enabled and that the passwords in UCM and Application IPPM are the same, also make sure the IPPM status is set to on. Additionally if you want to see presence updates make sure you have your SIP trunk from UCM to CUPS set properly and that the user that you want to see presence updates from has been associated with the line/DN. 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] [OSL|CCIE Voice] no notification sound on ip phone for messages
the authentication URL in enterprise parameters is so that the user can log into the IPPM service. If you are logged in then you have correctly setup the enterprise parameter. You probably did it by default when you changed the url from DNS based to IP based. Meaning you changed CUCMPUB to 10.10.210.10 within the URL. From: Krishna [vinayak_...@yahoo.com] Sent: Monday, July 23, 2012 5:59 PM To: Bruno Nonogaki; Justin McIntyre Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] [OSL|CCIE Voice] no notification sound on ip phone for messages Bruno Justin, I did enabled all these and even verified twice... what authentication url has to be there on enterprise parameters??/ thank you krishna. From: Bruno Nonogaki brun...@gmail.com To: Justin McIntyre justin.mcint...@blackbox.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Monday, July 23, 2012 4:39 PM Subject: Re: [OSL | CCIE_Voice] [OSL|CCIE Voice] no notification sound on ip phone for messages And also check the Authenticate URL on Enterprise Parameters... On Mon, Jul 23, 2012 at 5:44 PM, Justin McIntyre justin.mcint...@blackbox.commailto:justin.mcint...@blackbox.com wrote: this means you do not have all configuration completed. You need to check these few places: 1. user licensed for CUP in UCM 2. You have created the Application user for IPPM(PhoneMessenger) in UCM and the phone you are using the IPPM service on is associated with this Application user.. Also make sure this user is CTI enabled and that the passwords in UCM and Application IPPM are the same, also make sure the IPPM status is set to on. Additionally if you want to see presence updates make sure you have your SIP trunk from UCM to CUPS set properly and that the user that you want to see presence updates from has been associated with the line/DN. 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.comhttp://www.ipexpert.com/ Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ 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] [OSL | CCIE_VOICE] CUC imported users behaviour
This is Bug CSCsw83747 Go to the CUC 7.0(2) release notes and search for the
bug..
http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/release/notes/702cucrn.html
From: ccie_voice-boun...@onlinestudylist.com
[ccie_voice-boun...@onlinestudylist.com] On Behalf Of
ccie_voice-requ...@onlinestudylist.com [ccie_voice-requ...@onlinestudylist.com]
Sent: Thursday, July 12, 2012 12:00 PM
To: ccie_voice@onlinestudylist.com
Subject: CCIE_Voice Digest, Vol 77, Issue 19
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: CUC imported users behaviour (Kevin Spicer)
2. Overlapping route patterns (The Masterplan)
3. Re: Overlapping route patterns (Dan Quinlan (daquinla))
--
Message: 1
Date: Wed, 11 Jul 2012 19:02:55 +0100
From: Kevin Spicer ke...@kevinspicer.co.uk
To: The Masterplan winmasterp...@gmail.com
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CUC imported users behaviour
Message-ID:
caf2ggoqrtl602iv-vun-57bqw8jt5rgedg8p1xbeccare4+...@mail.gmail.com
Content-Type: text/plain; charset=iso-8859-1
I don't have it to hand, that was from memory :( but if you search.the
archive for those commands I have posted it before.
On 11 Jul 2012 18:41, The Masterplan winmasterp...@gmail.com wrote:
Awesome Kevin. I got burned 1 day by this bug. Can you give me the cisco
troubleshoot link from where you get those commands?
Thank you once again.
On Wed, Jul 11, 2012 at 8:17 PM, Kevin Spicer ke...@kevinspicer.co.ukwrote:
There's a bug in cuc that prevents mail from pstn being delivered if the
smtp domain has been changed. Perhaps you hit that (affects the 5 lab
workbook lab rentals).
run cuc dbquery unitydirdb select * from tbl_alias
to see if domains all match.
If not fix with
run cuc dbquery unitydirdb EXECUTE PROCEDURE csp_SmtpAddressMigrate('
new.com','old.com')
(substitute the correct old and new domain)
On 11 Jul 2012 18:03, The Masterplan winmasterp...@gmail.com wrote:
Hi,
I am struggling with a strange CUC behaviour and I am interested if
anyone experienced this.
For users imported from CUC, no message is being recorded in user
mailbox, altough the system play (after the user press # for more options
and then 1 to send this message ) your message has been sent. On that
imported user, if I go to edit- Mailbox it shows that the number of
messages is 0.
If I create users from CUC administration everything is working fine, I
receive MWI and I see the corresponding number of messages in user mailbox.
In order to solve this issue, I deactivated/activated all the unity
services and then I restarted CUCM and CUC and still nothing. I usually
activate all the CUC services and then I start to configure it (Phone
System, Port group, Port, User templates etc) and in this case I've done
the same.
Thank you experts,
___
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
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Message: 2
Date: Thu, 12 Jul 2012 13:51:27 +0300
From: The Masterplan winmasterp...@gmail.com
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Overlapping route patterns
Message-ID:
CAKsL8KB9k2=w5efs8yoww6ghsyojnt0+yiatitjouqjrwso...@mail.gmail.com
Content-Type: text/plain; charset=iso-8859-1
Hi,
I have the following route patterns on CUCM that have a mgcp gateway in
route list:
911 - marked as urgent priority
9.XXX - discard digit predot
91112.XXX- discard digit predot
Because those are overlapping route patterns, their behaviour is different
depending if enbloc dialing is used or not:
1)With digit by digit dialing if I dial 923942123 it will go to 911
because this route pattern have urgent priority enabled. If I dial 91112345
it will also go to 911 route pattern.
2)With en bloc dialing if I dial 923942123 it will go to the third
route pattern as desired. If I dial 91112345 it will go to the second route
pattern.
So en bloc dialing will work good in this scenario. But if the routes above
are configured in a CUCME, as dial-peers, I cannot find a way to
[OSL | CCIE_Voice] (no subject)
the trick here is in your patterns digit analysis. You need the put the
expected digits to seperate LD from say Local or international
try this out and see what happens ...
911 - marked as urgent priority
9.[2-9]XX - discard digit predot
90112.XXX- discard digit predot
and of course you can change this how you wan to but you need to differentiate
betwwen Local, LD and Int dialing. The way to do this is with the prepending
digits...so for NANP dialing you would have..
911 urgent
9.[2-9]XX local 7 digit
9.1[2-9]..[2-9].. LD 11 digit (10 digit will always work)
9.011! international pattern
now enblock or digit by digit analysis will succeed.
Message: 2
Date: Thu, 12 Jul 2012 13:51:27 +0300
From: The Masterplan winmasterp...@gmail.com
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Overlapping route patterns
Message-ID:
CAKsL8KB9k2=w5efs8yoww6ghsyojnt0+yiatitjouqjrwso...@mail.gmail.com
Content-Type: text/plain; charset=iso-8859-1
Hi,
I have the following route patterns on CUCM that have a mgcp gateway in
route list:
911 - marked as urgent priority
9.XXX - discard digit predot
91112.XXX- discard digit predot
Because those are overlapping route patterns, their behaviour is different
depending if enbloc dialing is used or not:
From: ccie_voice-boun...@onlinestudylist.com
[ccie_voice-boun...@onlinestudylist.com] On Behalf Of
ccie_voice-requ...@onlinestudylist.com [ccie_voice-requ...@onlinestudylist.com]
Sent: Thursday, July 12, 2012 12:00 PM
To: ccie_voice@onlinestudylist.com
Subject: CCIE_Voice Digest, Vol 77, Issue 19
Send CCIE_Voice mailing list submissions to
ccie_voice@onlinestudylist.com
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than Re: Contents of CCIE_Voice digest...
Today's Topics:
1. Re: CUC imported users behaviour (Kevin Spicer)
2. Overlapping route patterns (The Masterplan)
3. Re: Overlapping route patterns (Dan Quinlan (daquinla))
--
Message: 1
Date: Wed, 11 Jul 2012 19:02:55 +0100
From: Kevin Spicer ke...@kevinspicer.co.uk
To: The Masterplan winmasterp...@gmail.com
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CUC imported users behaviour
Message-ID:
caf2ggoqrtl602iv-vun-57bqw8jt5rgedg8p1xbeccare4+...@mail.gmail.com
Content-Type: text/plain; charset=iso-8859-1
I don't have it to hand, that was from memory :( but if you search.the
archive for those commands I have posted it before.
On 11 Jul 2012 18:41, The Masterplan winmasterp...@gmail.com wrote:
Awesome Kevin. I got burned 1 day by this bug. Can you give me the cisco
troubleshoot link from where you get those commands?
Thank you once again.
On Wed, Jul 11, 2012 at 8:17 PM, Kevin Spicer ke...@kevinspicer.co.ukwrote:
There's a bug in cuc that prevents mail from pstn being delivered if the
smtp domain has been changed. Perhaps you hit that (affects the 5 lab
workbook lab rentals).
run cuc dbquery unitydirdb select * from tbl_alias
to see if domains all match.
If not fix with
run cuc dbquery unitydirdb EXECUTE PROCEDURE csp_SmtpAddressMigrate('
new.com','old.com')
(substitute the correct old and new domain)
On 11 Jul 2012 18:03, The Masterplan winmasterp...@gmail.com wrote:
Hi,
I am struggling with a strange CUC behaviour and I am interested if
anyone experienced this.
For users imported from CUC, no message is being recorded in user
mailbox, altough the system play (after the user press # for more options
and then 1 to send this message ) your message has been sent. On that
imported user, if I go to edit- Mailbox it shows that the number of
messages is 0.
If I create users from CUC administration everything is working fine, I
receive MWI and I see the corresponding number of messages in user mailbox.
In order to solve this issue, I deactivated/activated all the unity
services and then I restarted CUCM and CUC and still nothing. I usually
activate all the CUC services and then I start to configure it (Phone
System, Port group, Port, User templates etc) and in this case I've done
the same.
Thank you experts,
___
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please visit www.ipexpert.com
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-- next
[OSL | CCIE_Voice] Time sync issue with 7961s
So I think there could be a couple different things to check here 1. What protocol are the 7961s running? 7960 protocol? If you're running a mixture of SIP and SCCP on these phones you will need to setup a NTP reference for the device pool to which the SIP phones are part of. This also leads me to ask if you have synced your PUB server to the CME router as well? If, (and I'm only assuming here) you have SIP phones that do not have a NTP reference setup, then they will get their time update from the final 200 ok message they receive from the PUB/SUB server during registration. Ergo... if PUB isn't synchronized to CME then the SUB (which automatically syncs to the PUB for timing) will be handing out an erroneous time stamp in the final 200 ok message to the SIP phone. This is of course assuming you are using SIP phones. 2. What kind of IP scheme do you have setup? I would assume that your behind a NATed firewall of somekind (home wifi router etc.) so to the NTP server he should only see the requests coming from one ip address. 3. Since you say you are syncing the phones to the CME router... Did you configure your CME router to source NTP packets from a certain ip address within the CME router? 4. Run a debug ntp all and then reset your phones. Check the output to see what and where the messages are actually doing and going. Just some thoughts, but If this is not the scenario you are facing or does not help you in any way then shoot us a running-config of the CME router as well as a show ntp status and show ntp associations. And we'll take a closer look and get you going. Hope this helps have a great day! Thanks, Justin McIntyre 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
[OSL | CCIE_Voice] OSL | CCIE_Voice] Time sync issue with 7961s
My apologies. I totally read past one key word CME sorry about that. Well none the less , send us a running-config and myself and hopefully someone that doesn't miss the forrest for the trees will be able to help. Thanks, Justin McIntyre 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
[OSL | CCIE_Voice] MWI Issues with CUE
Back in March I had some issues with MWI inter-workings when using JTAPI integration to CUCM and having SRST as a backup. I kept noticing that though my configurations within CUE and the gateway IOS seemed correct I just couldn't get the unsolicited Notify messages to propagate over to the gateway from CUE. Well last night I had a break through. I may be the only one that this has ever happened to but if this helps anyone else out there then I will consider it a success. I had my SIP stack IP set to the same ip that SIP was bound to in the gateway IOS and Sip-UA set to unsolicited but, the issue that I was facing had to do with the integration between CUE and CUCM not the SIP configurations within the gateway IOS. When integrating CUE with CUCM, out of habit , I would always list the primary and backup UCM servers during setup. Whenever I would simulate a failover I would put my BR2/Site C users in a Subscriber only Device Pool and then stop the Call manager Service on the Subscriber server. The issue here is that CUE will failover to the Publisher server instead of failing over to an SRST type of functionality. Therefore when an incoming call for BR2 phones came into the gateway, the active dial-peer would point to the locally registered phone which would in turn push a call-forward busy/no answer to the CUE module. A message would be left and then instead of a unsolicited MWI notification being sent out via the SIP stack, the CUE modul e would send the MWI update out of the CTI port that was currently registered with the PUB server. By changing the CUE to only the subscriber server for the JTAPI integration and resetting the CUE, once the call manager service on the Subscriber is stopped, then the CUE module will not have another server to failover to and the SIP MWI messages will then be sourced from the CUE SIP stack. Note: Using the ip route pub.ip.address.x 255.255.255.255 null0 commands will also work here; if you do it for both the SUB and the PUB. This method will kill connectivity directly to the PUB and SUB servers which will do the trick as well. Like I said, I may be the only one that this has bitten, but if there's one other person that this helps then great! Thanks, Justin 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
[OSL | CCIE_Voice] class-based cRTP
drops) 0/0/0 (pkts output/bytes output) 3525/229878 shape (average) cir 384000, bc 9600, be 0 target shape rate 384000 lower bound cir 0, adapt to fecn 0 Service-policy : LLQ queue stats for all priority classes: Queueing queue limit 64 packets (queue depth/total drops/no-buffer drops) 0/0/0 (pkts output/bytes output) 147/7512 Class-map: SIG (match-any) 147 packets, 7512 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: protocol skinny 147 packets, 7512 bytes 5 minute rate 0 bps Match: protocol h323 0 packets, 0 bytes 5 minute rate 0 bps Match: protocol mgcp 0 packets, 0 bytes 5 minute rate 0 bps Match: protocol sip 0 packets, 0 bytes 5 minute rate 0 bps Match: protocol rsvp 0 packets, 0 bytes 5 minute rate 0 bps Priority: 18 kbps, burst bytes 1500, b/w exceed drops: 0 Class-map: RTP (match-any) 3257 packets, 208448 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: access-group 101 3257 packets, 208448 bytes 5 minute rate 0 bps Priority: 24 kbps, burst bytes 1500, b/w exceed drops: 0 compress: header ip rtp UDP/RTP (compression on, Cisco, RTP) Sent:0 total, 0 compressed, 0 bytes saved, 0 bytes sent rate 0 bps Class-map: class-default (match-any) 120 packets, 13854 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: any Queueing queue limit 64 packets (queue depth/total drops/no-buffer drops/flowdrops) 0/0/0/0 (pkts output/bytes output) 121/13918 Fair-queue: per-flow queue limit 16 Thanks, Justin McIntyre 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] CCIE_Voice Digest, Vol 76, Issue 50
Krishna, I don't think your quite to the Media termination stage yet. You must first get out of the Ringing or signaling portion of the call setup. That being said, the Wait for Far end TCS (terminal capabilities set) is not supported when using CUBE. Go to your trunk pointing to Gatekeeper and uncheck this mark. Keep in mind that depending on what codecs you are using or if you require a Media Termination Point (MTP) for any reason you may need to set that up here. Since you are going to use the Transcoder at the CUBE gateway, you can just send the call out as G711u/a from CUCM to the Gatekeeper/CUBE device. With the Wait for far end TCS unchecked, there will no longer be a stalemate when the H323 signaling transitions to the setup stage. Someone please correct me here if I'm wrong. Note: you may want to check that your hold functionality still works. You may see that once the call is connected if you put the user on hold, you may not beable to retrieve the call from hold again. At this point you will need an MTP to facilitate TCS functionality, if you are indeed using G711u/a from CUCM then use the build in MTPs, if not, then setup an MTP on your local IOS gateway. ***Someone please correct me if I'm leading Krishna astray. Thanks, Justin McIntyre 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
[OSL | CCIE_Voice] LAB8 - Question 7.1
You could try using number expansion. Additionally, you can apply number expansion via a regular expression to either the Serial Interface or the Voice-Port I can't remember which. Technically this is not a voice translation-rule. Thanks, Justin McIntyre 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] LAB8 - Question 7.1
I may have a chance to look it up here in a bit. But in the Volume 1 labs , if you have them, Vik goes over a command on how to put a regular expression into config within either the voice-port or serial port (I can't remember which). I'll look into it here in a bit and try to send an update. I could be way off, I'm sure that I am, but I could have swore there was some manipulation that could be done at one of those locations. I just can't remember the details right now but figured it might be a good place to start looking. I'll update when I can. Thanks, Justin McIntyre From: Bruno Nonogaki [mailto:brun...@gmail.com] Sent: Sunday, June 17, 2012 11:27 AM To: Justin McIntyre Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] LAB8 - Question 7.1 Hey guys, Yeah, maybe num-exp and dialplan-pattern could be used for inbound calls. To translate the number received from PSTN to 4 digits extension. But it can't manipulate the ANI to 10 Digits on outgoing calls. I would still need voice translation-rules. And what do you mean by a number expansion on Serial or voice-port? Do you mean the regular translation-rule X instead of voice translation-rules / voice translation-profile? If so, the question says you can't use both... :( Thank you, Bruno On Sun, Jun 17, 2012 at 9:35 AM, Justin McIntyre justin.mcint...@blackbox.commailto:justin.mcint...@blackbox.com wrote: You could try using number expansion. Additionally, you can apply number expansion via a regular expression to either the Serial Interface or the Voice-Port I can't remember which. Technically this is not a voice translation-rule. Thanks, Justin McIntyre 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.comhttp://www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.PlatinumPlacement.com No virus found in this message. Checked by AVG - www.avg.comhttp://www.avg.com Version: 2012.0.1913 / Virus Database: 2433/5075 - Release Date: 06/17/12 ___ 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] IPPM Problem
Should be a service parameter for remembering last login. Thanks, Justin McIntyre Engineer 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] CCIE_Voice Digest, Vol 76, Issue 42
Randall is correct here. UCM will always divert to the intra-region Service Parameter settings. Change this to 729 and then hard code your codec between regions within the region parameters section. Thanks, Justin McIntyre Engineer Mutual Telecom Services Inc. a wholly-owned subsidiary of Black Box Corp. COMM: (434)-946-1562 DSN: (312)-237-1562 CELL: (540)-312-9391 FAX: (434)-946-1510 -Original Message- Message: 6 Date: Thu, 14 Jun 2012 00:14:03 -0700 From: Rrcrumm rrcr...@yahoo.com To: Krishna vinayak_...@yahoo.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Gatekeeper acting weird for codec Message-ID: 0d193090-0965-4053-9d15-0a2d160ea...@yahoo.com Content-Type: text/plain; charset=us-ascii I think I remember some tone a k this situation. You chance the setting is service parameters for intra region to g729 HTH Randall Sent from my iPhone On Jun 13, 2012, at 9:10 PM, Krishna vinayak_...@yahoo.com wrote: Hi folks, I configured the gatekeeper on the Hq router, and when i call from hq to br2(cme) the call set up shows as 16 kbps, but whereas from cme to hq it shows as 128 kbps but the actual call is connected with g729. Even after the call got connected, it stills shows as 128 in the show gatekeeper call. here is the output status: i put the gk-trunk in the hq region as well. any help is much appreciated on this matter. BR2-RTR(config)#do sh voice call stat CallID CID ccVdb Port DSP/Ch Called # Codec Dial-peers 0xC0 13B3 0x49DA073C 50/0/2.0 95002 g729r8 20002/15 1 active call found HQ-RTR(config-gk)#do sh gatek call Total number of active calls = 1. largest hash bucket = 1 GATEKEEPER CALL INFO LocalCallIDAge(secs) BW 62-4230757933 128(Kbps) ConferenceID CallID SrcCRV A5CF2B43 B52E11E1 81D2B06F 708A730B A5CF2B43 B52E11E1 81D4B06F 708A730B 85 Endpt(s): Alias E.164Addr src EP: BR2-RTR 3002 CallSignalAddr Port RASSignalAddr Port 10.10.110.3 1820 10.10.110.3 62007 Endpt(s): Alias E.164Addr dst EP: gk-trunk_195002 CallSignalAddr Port RASSignalAddr Port 10.10.210.101720 10.10.210.1032784 callstate: SEP, DEP, 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 -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20120614/7325e5c0/attachment.html -- ___ CCIE_Voice mailing list CCIE_Voice@onlinestudylist.com http://onlinestudylist.com/mailman/listinfo/ccie_voice End of CCIE_Voice Digest, Vol 76, Issue 42 ** 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
[OSL | CCIE_Voice] DTMF from BR1 phone to CUC via SIP trunk notrecognized
Depending on the situation you could use bothif warranted of course. Lets say that HQ is uncompressed to BR2 but BR1 does use cRTP. What if you wanted to continue to use RFC2833 at HQ site? You could simply choose the both DTMF option on the SIP trunk settings page and then Disable RFC2833 for the BR1 phones via the Phones Admin page. This way HQ phones would continue to use 2833 and BR1, which has a compressed path, will use OOB. Just a thought. Thanks, Justin McIntyre -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Tuesday, June 12, 2012 1:01 PM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 76, Issue 39 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: DTMF from BR1 phone to CUC via SIP trunk notrecognized properly (Bill Lake) 2. Re: DTMF from BR1 phone to CUC via SIP trunk notrecognized properly (Tapan Gautam (tgautam)) 3. Re: DTMF from BR1 phone to CUC via SIP trunk notrecognized properly (Bill Lake) -- Message: 1 Date: Tue, 12 Jun 2012 11:29:18 -0500 From: Bill Lake whl...@gmail.com To: Krishna vinayak_...@yahoo.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] DTMF from BR1 phone to CUC via SIP trunk notrecognized properly Message-ID: cadpb93o93ds0xjqfaks8ghxzfxmujjo3upa1pi+e1qnksnw...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Might want to read this as it is the authority on RFC 2833 and it appears to be out of band however it uses the RTP header. It also appears that the tones are actually carried in the RTP traffic just not the audio stream. http://www.ietf.org/rfc/rfc2833.txt Would it be interesting to see if cRTP was turned off if DTMF would work in this case? On Tue, Jun 12, 2012 at 7:58 AM, Krishna vinayak_...@yahoo.com wrote: Dan, A small correction to your statement..rfc2833 is out of band mechanism mostly, and moreover it doesn't use audio channel, infact it uses rtp header to relay the dtmf message with a payload identifier. thank you Krishna.. -- *From:* Dan Quinlan (daquinla) daqui...@cisco.com *To:* Tapan Gautam (tgautam) tgau...@cisco.com *Cc:* ccie_voice@onlinestudylist.com *Sent:* Monday, June 11, 2012 11:57 PM *Subject:* Re: [OSL | CCIE_Voice] DTMF from BR1 phone to CUC via SIP trunk notrecognized properly cRTP mangles in-band (audio) DTMF. If I understand correctly, you are SIP-integrated between CUC and UCM. You need to be OOB only for DTMF (not rfc2833). Rfc2833 is an in-band (audio channel) mechanism. You need OOB (signaling channel) for DTMF to function when cRTP is used. DQ d...@cisco.com Sent from my iPhone On Jun 11, 2012, at 11:33 PM, Tapan Gautam (tgautam) tgau...@cisco.com wrote: 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) ? g729r8 with crtp ? CUCM ? SIP trunk(with OOB and RFC2833 as dtmf options) ? 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 ___ 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 -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20120612/247d7cb2/attachment-0001.html -- Message
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] PSTN issues
What is the phone type? You might try doing a hard reset of the phone it's self. Press and hold # while you power cycle, when lights start flashing input 123456789*0# let it sit. Note make sure you have your dhcp and TFTP server still reachable at this point. Just wait and let the phone do its thing. I have seen in the past where some of my phones act a little screwy like this every now and again, sometimes this reset straightens out. Thanks, Justin McIntyre -Original Message- Message: 3 Date: Fri, 8 Jun 2012 04:56:40 + From: Leslie Meade leslie.me...@lvs1.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] PSTN issues Message-ID: f64719604b4e6f41bdbb2af38e7609f4181f3...@lvscgyex03.longviewsystems.com Content-Type: text/plain; charset=iso-8859-1 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 1deviceType 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
Re: [OSL | CCIE_Voice] hunt group via AA not working
Is there any way we could see the rest of your config? Where is the Hunt group configured? Are the phones running in CME,UCM? I have seen it once and it was due to the way I configured the hunt group in CME. I used hunt-group instead of Voice hunt-group and I think that's what caused it to break going through the BACD AA application. Just a thought. Thanks, Justin Message: 4 Date: Tue, 22 May 2012 20:28:54 -0700 (PDT) From: Krishna vinayak_...@yahoo.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] hunt group via AA not working Message-ID: 1337743734.93075.yahoomail...@web46001.mail.sp1.yahoo.com Content-Type: text/plain; charset=iso-8859-1 Folks, it is so strange that when i call the hunt group number 3210 from pstn, both sip and sccp phone rings. But with AA on cme, only cisco phone rings but not both. even i verified with the config, and i see hunt group as the right option when user presses the digit 2. does anyone know why this is happening only for AA??? here is the config: application ?service queue flash:bacdprompts/app-b-acd-2.1.2.2.tcl ? param number-of-hunt-grps 2 ? param aa-hunt2 3210 ? param aa-hunt10 3006 ? param queue-len 15 ? param queue-manager-debugs 1 service aa flash:bacdprompts/app-b-acd-aa-2.1.2.2.tcl ? paramspace english index 1 ? paramspace english language en ? paramspace english location flash:bacdprompts/ ? param service-name queue ? param handoff-string aa ? param aa-pilot 3500 ? param welcome-prompt _bacd_welcome.au ? param number-of-hunt-grps 2 ? 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 3001 -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20120522/47dc2af6/attachment.html -- __ 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] hunt group via AA not working
There it is . I couldn't remember which one it supported. Thanks, Justin McIntyre Engineer Mutual Telecom Services Inc. a wholly-owned subsidiary of Black Box Corp. COMM: (434)-946-1562 DSN: (312)-237-1562 CELL: (540)-312-9391 FAX: (434)-946-1510 -Original Message- From: Mohd Baqari [mailto:baqari.voic...@gmail.com] Sent: Wednesday, May 23, 2012 3:03 PM To: Justin McIntyre Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] hunt group via AA not working Plz share the full config including hunt groups. I remeber that bacd supports ephone hunt but not voice hint groups. Regards, Mohammed Al Baqari Sent from my iPhone On May 23, 2012, at 4:06 PM, Justin McIntyre justin.mcint...@blackbox.com wrote: Is there any way we could see the rest of your config? Where is the Hunt group configured? Are the phones running in CME,UCM? I have seen it once and it was due to the way I configured the hunt group in CME. I used hunt-group instead of Voice hunt-group and I think that's what caused it to break going through the BACD AA application. Just a thought. Thanks, Justin Message: 4 Date: Tue, 22 May 2012 20:28:54 -0700 (PDT) From: Krishna vinayak_...@yahoo.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] hunt group via AA not working Message-ID: 1337743734.93075.yahoomail...@web46001.mail.sp1.yahoo.com Content-Type: text/plain; charset=iso-8859-1 Folks, it is so strange that when i call the hunt group number 3210 from pstn, both sip and sccp phone rings. But with AA on cme, only cisco phone rings but not both. even i verified with the config, and i see hunt group as the right option when user presses the digit 2. does anyone know why this is happening only for AA??? here is the config: application ?service queue flash:bacdprompts/app-b-acd-2.1.2.2.tcl ? param number-of-hunt-grps 2 ? param aa-hunt2 3210 ? param aa-hunt10 3006 ? param queue-len 15 ? param queue-manager-debugs 1 service aa flash:bacdprompts/app-b-acd-aa-2.1.2.2.tcl ? paramspace english index 1 ? paramspace english language en ? paramspace english location flash:bacdprompts/ ? param service-name queue ? param handoff-string aa ? param aa-pilot 3500 ? param welcome-prompt _bacd_welcome.au ? param number-of-hunt-grps 2 ? 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 3001 -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20120522/47dc2af6/attachment.html -- __ 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 - No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.1913 / Virus Database: 2425/5017 - Release Date: 05/23/12 ___ 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] unity connection 8
I have it installed and resources was my issue. The first thing I would do is look for the OVA/OVF for UCxn 8.x on Cisco's website. Look at the resources required and match your VM settings accordingly. I know that Unity Connection requires an additional VCPU to be taken up to allocate for scheduling so It may be that they get you to utilize it by assigning an extra one to the VM. Personally the first thing I would do it create a VM with 2 VCPU, 8 gig of ram and 250 gig drive(this is overkill but if it works you can scale it back from there). If it still doesn't allow you to install it then I'd go from there but I'm almost 90% positive this is the issue your facing. When you install the application, the installer runs a check to see if your system (VM) can support the applications available on that disk. At the time of application selection only those apps that can be supported by your currently setup VM are shown. Bump the resources up a bit more and see what happen s from there. Then only other thing that may be getting you would possibly be CPU power, but I doubt it. If you have a Dell 1950 II then I'd think you're doing ok, especially if you have the Dual Quad core model, but even still it should work. Thanks, Justin McIntyre 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
[OSL | CCIE_Voice] [OSL | CCIE_ Voice] Single Number Reach/ MVA
Hello all, I know that it's best practice to use Complete Match when using MVA/SNR in the 7.0 CUCM load but i have a question on how to get around something. I'm working in the 5 labs and can't find a way to keep complete match becuase of how the PSTN switch is sending ANI to the gateways. Scenario: line 2 of pstn is Remote Destination of HQ PHN 2 line 2002. Line 2 of PSTN phone (2024678124) calls 2025552001 The ANI/DNIS sent from PSTN to HQ Gateway is ... Calling Party Number i = 0x2180, '4678124' Plan:ISDN, Type:Subscriber(local) Called Party Number i = 0xA1, '2025553001' Line 2 of PSTN phone (2024678124) calls 4083873001 The ANI/DNIS information sent to Branch 1 gateway is... Calling Party Number i = 0x2180, '2024678124' Plan:ISDN, Type:National Called Party Number i = 0xA1, '4083873001' As you can see that depending on which location i make a call to the PSTN formats the number of digits as well as plan and type accordingly. Is there a way to get around not having to use partial match so that the end user will always see the call as from 2002 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
[OSL | CCIE_Voice] [OSL | CCIE_VOICE] 5 Labs , lab 3, 4.4
I have a question about GK and ACF messaging. In the latter part of the question 4.4 we are going over the bandwidth requests that are used during the ARQ and ACF messaging. CME sends a request to the Gatekeeper based upon the default codec setting of G729 in the outbound Dial Peer. It request 16kbps for the call. When answering the call, UCM requests 128kbps from Gatekeeper based upon its intra region codec setting. I understand why and how we should change the intra region codec setting in UCM service parameters to make sure that we do not falsely allocate bandwidth that is not being utilized from the Gatekeeper in the event of Gatekeeper CAC. My question is how doe the BRQ service parameter fall in with this? My original interpretation is that when the first request is made for higher bandwidth request that upon answering the call with a lower codec rate that the bandwidth request would then be re-initiated to reflect the proper lower rate.. Thanks in advance for clarification. Thanks, Justin McIntyre Engineer Mutual Telecom Services Inc. a wholly-owned subsidiary of Black Box Corp. COMM: (434)-946-1562 DSN: (312)-237-1562 CELL: (540)-312-9391 FAX: (434)-946-1510 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] [OSL | CCIE_VOICE] SRST Auto Provision All
Thank you very much for that article. I hadn't done that yet cause I really
wanted to get the dial-peer to correctly transition to down. None the less I
also have CFUR configured for those devices. This will create an endless loop
if I am correct? But I also know that setting the Max Forward UnRegistered
Hops to DN javascript:getHelp('MaxForwardUnRegisteredsToDn') Service
parameter. So the Dial
Per hunt 2 command followed by the previously listed service parameter should
take care of this issue.
Thanks again for the link and all those who helped.
Thanks,
Justin McIntyre
Engineer
Mutual Telecom Services Inc.
a wholly-owned subsidiary of Black Box Corp.
COMM: (434)-946-1562
DSN: (312)-237-1562
CELL: (540)-312-9391
FAX: (434)-946-1510
[cid:image001.gif@01CD0E51.C0B0BB40][cid:image002.gif@01CD0E51.C0B0BB40][cid:image003.gif@01CD0E51.C0B0BB40]
Please note new e-mail address
justin.mcint...@blackbox.commailto:justin.mcint...@blackbox.com
From: Eliot Ngwa [mailto:eliot.n...@gmail.com]
Sent: Thursday, March 29, 2012 11:20 PM
To: Justin McIntyre
Cc: Bill Lake; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] [OSL | CCIE_VOICE] SRST Auto Provision All
Check out this link. It's similar to what you're facing with a proper
workaround:
http://blog.ipexpert.com/2012/03/12/high-availability-series-3-unified-cme-for-srst-gotchas/
On Thu, Mar 29, 2012 at 10:34 PM, Justin McIntyre
justin.mcint...@blackbox.commailto:justin.mcint...@blackbox.com wrote:
Well it was supposed to learn them , but when it's the routes not in srst mode,
ie the devices are not registered to the router, the dial peers should
transition to down and they are not transitioning to down, ever after no
telephony-service and reload and rebuild. I know it's a known bug, but what I
thought would fix the bug is not working.
Thanks,
Justin McIntyre
Engineer
Mutual Telecom Services Inc.
a wholly-owned subsidiary of Black Box Corp.
COMM: (434)-946-1562tel:%28434%29-946-1562
DSN: (312)-237-1562tel:%28312%29-237-1562
CELL: (540)-312-9391tel:%28540%29-312-9391
FAX: (434)-946-1510tel:%28434%29-946-1510
[cid:image001.gif@01CD0E51.C0B0BB40][cid:image002.gif@01CD0E51.C0B0BB40][cid:image003.gif@01CD0E51.C0B0BB40]
Please note new e-mail address
justin.mcint...@blackbox.commailto:justin.mcint...@blackbox.com
From: Bill Lake [mailto:whl...@gmail.commailto:whl...@gmail.com]
Sent: Thursday, March 29, 2012 10:32 PM
To: Justin McIntyre
Subject: Re: [OSL | CCIE_Voice] [OSL | CCIE_VOICE] SRST Auto Provision All
Could be it learned them there and how do you delete that?
On Thu, Mar 29, 2012 at 9:15 PM, Justin McIntyre
justin.mcint...@blackbox.commailto:justin.mcint...@blackbox.com wrote:
Yes sorry i left that out
Thanks,
Justin McIntyre
Engineer
Mutual Telecom Services Inc.
a wholly-owned subsidiary of Black Box Corp.
COMM: (434)-946-1562tel:%28434%29-946-1562
DSN: (312)-237-1562tel:%28312%29-237-1562
CELL: (540)-312-9391tel:%28540%29-312-9391
FAX: (434)-946-1510tel:%28434%29-946-1510
Error! Filename not specified.Error! Filename not specified.Error! Filename not
specified.
Please note new e-mail address
justin.mcint...@blackbox.commailto:justin.mcint...@blackbox.com
From: Bill [mailto:whl...@gmail.commailto:whl...@gmail.com]
Sent: Thursday, March 29, 2012 10:01 PM
To: Justin McIntyre
Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] [OSL | CCIE_VOICE] SRST Auto Provision All
Was any srst provisioning done?
Bill
On Mar 29, 2012, at 7:15 PM, Justin McIntyre
justin.mcint...@blackbox.commailto:justin.mcint...@blackbox.com wrote:
Can anyone tell me why my learned ephone DNs would stay up even if I've removed
telephony-service, saved config, reloaded and then re-entered telephony service
back in. I also have some extra dial-peers showing up that I do not
reckognize. See output of show dial-peer voice summary below.
20001 pots up up 4001$ 0
50/0/1
20002 pots up up 4002$ 0
50/0/2
20003 pots up up 4000$ 0
50/0/3
20004 pots up up A01$ 0
50/0/10
20005 pots down down A4555A 0
50/0/1
20006 pots up up 4555 0
50/0/0
20007 pots down down A4555A0001 0
50/0/2
20008 pots down down A4555A0002 0
50/0/3
Thanks,
Justin McIntyre
Engineer
Mutual Telecom Services Inc.
a wholly-owned subsidiary of Black Box Corp.
COMM: (434)-946-1562tel:%28434%29-946-1562
DSN: (312)-237-1562tel:%28312%29-237-1562
CELL: (540)-312-9391tel:%28540%29-312-9391
FAX: (434)-946-1510tel:%28434%29-946-1510
image001.gifimage002.gifimage003.gif
Please note new e
[OSL | CCIE_Voice] [OSL | CCIE_VOICE] SRST Auto Provision All
Can anyone tell me why my learned ephone DNs would stay up even if I've removed telephony-service, saved config, reloaded and then re-entered telephony service back in. I also have some extra dial-peers showing up that I do not reckognize. See output of show dial-peer voice summary below. 20001 pots up up 4001$ 0 50/0/1 20002 pots up up 4002$ 0 50/0/2 20003 pots up up 4000$ 0 50/0/3 20004 pots up up A01$ 0 50/0/10 20005 pots down down A4555A 0 50/0/1 20006 pots up up 4555 0 50/0/0 20007 pots down down A4555A0001 0 50/0/2 20008 pots down down A4555A0002 0 50/0/3 Thanks, Justin McIntyre Engineer Mutual Telecom Services Inc. a wholly-owned subsidiary of Black Box Corp. COMM: (434)-946-1562 DSN: (312)-237-1562 CELL: (540)-312-9391 FAX: (434)-946-1510 [cid:image001.gif@01CD0DE8.A1F27E50][cid:image002.gif@01CD0DE8.A1F27E50][cid:image003.gif@01CD0DE8.A1F27E50] Please note new e-mail address justin.mcint...@blackbox.commailto:justin.mcint...@blackbox.com 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. inline: image001.gifinline: image002.gifinline: image003.gif___ 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] [OSL | CCIE_VOICE] SRST Auto Provision All
Well it was supposed to learn them , but when it's the routes not in srst mode, ie the devices are not registered to the router, the dial peers should transition to down and they are not transitioning to down, ever after no telephony-service and reload and rebuild. I know it's a known bug, but what I thought would fix the bug is not working. Thanks, Justin McIntyre Engineer Mutual Telecom Services Inc. a wholly-owned subsidiary of Black Box Corp. COMM: (434)-946-1562 DSN: (312)-237-1562 CELL: (540)-312-9391 FAX: (434)-946-1510 [cid:image001.gif@01CD0DFC.1B55F430][cid:image002.gif@01CD0DFC.1B55F430][cid:image003.gif@01CD0DFC.1B55F430] Please note new e-mail address justin.mcint...@blackbox.commailto:justin.mcint...@blackbox.com From: Bill Lake [mailto:whl...@gmail.com] Sent: Thursday, March 29, 2012 10:32 PM To: Justin McIntyre Subject: Re: [OSL | CCIE_Voice] [OSL | CCIE_VOICE] SRST Auto Provision All Could be it learned them there and how do you delete that? On Thu, Mar 29, 2012 at 9:15 PM, Justin McIntyre justin.mcint...@blackbox.commailto:justin.mcint...@blackbox.com wrote: Yes sorry i left that out Thanks, Justin McIntyre Engineer Mutual Telecom Services Inc. a wholly-owned subsidiary of Black Box Corp. COMM: (434)-946-1562tel:%28434%29-946-1562 DSN: (312)-237-1562tel:%28312%29-237-1562 CELL: (540)-312-9391tel:%28540%29-312-9391 FAX: (434)-946-1510tel:%28434%29-946-1510 Error! Filename not specified.Error! Filename not specified.Error! Filename not specified. Please note new e-mail address justin.mcint...@blackbox.commailto:justin.mcint...@blackbox.com From: Bill [mailto:whl...@gmail.commailto:whl...@gmail.com] Sent: Thursday, March 29, 2012 10:01 PM To: Justin McIntyre Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] [OSL | CCIE_VOICE] SRST Auto Provision All Was any srst provisioning done? Bill On Mar 29, 2012, at 7:15 PM, Justin McIntyre justin.mcint...@blackbox.commailto:justin.mcint...@blackbox.com wrote: Can anyone tell me why my learned ephone DNs would stay up even if I've removed telephony-service, saved config, reloaded and then re-entered telephony service back in. I also have some extra dial-peers showing up that I do not reckognize. See output of show dial-peer voice summary below. 20001 pots up up 4001$ 0 50/0/1 20002 pots up up 4002$ 0 50/0/2 20003 pots up up 4000$ 0 50/0/3 20004 pots up up A01$ 0 50/0/10 20005 pots down down A4555A 0 50/0/1 20006 pots up up 4555 0 50/0/0 20007 pots down down A4555A0001 0 50/0/2 20008 pots down down A4555A0002 0 50/0/3 Thanks, Justin McIntyre Engineer Mutual Telecom Services Inc. a wholly-owned subsidiary of Black Box Corp. COMM: (434)-946-1562tel:%28434%29-946-1562 DSN: (312)-237-1562tel:%28312%29-237-1562 CELL: (540)-312-9391tel:%28540%29-312-9391 FAX: (434)-946-1510tel:%28434%29-946-1510 image001.gifimage002.gifimage003.gif Please note new e-mail address justin.mcint...@blackbox.commailto:justin.mcint...@blackbox.com 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.comhttp://www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.PlatinumPlacement.com No virus found in this message. Checked by AVG - www.avg.comhttp://www.avg.com Version: 2012.0.1913 / Virus Database: 2114/4903 - Release Date: 03/29/12 No virus found in this message. Checked by AVG - www.avg.comhttp://www.avg.com Version: 2012.0.1913 / Virus Database: 2114/4903 - Release Date: 03/29/12 inline: image001.gifinline: image002.gifinline: image003.gif___ 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] Teamadmin access for UCCX7
I'm not sure what Desktop Administrator/Side A is but I can access the Admin utility through programsCiscodesktopadmin. Thanks, Justin McIntyre Engineer Mutual Telecom Services Inc. a wholly-owned subsidiary of Black Box Corp. COMM: (434)-946-1562 DSN: (312)-237-1562 CELL: (540)-312-9391 FAX: (434)-946-1510 [cid:image001.gif@01CD099B.D8B6EC60][cid:image002.gif@01CD099B.D8B6EC60][cid:image003.gif@01CD099B.D8B6EC60] Please note new e-mail address justin.mcint...@blackbox.commailto:justin.mcint...@blackbox.com From: Gurpreet Singh Kukreja [mailto:tycoononway1...@gmail.com] Sent: Saturday, March 24, 2012 8:51 AM To: Justin McIntyre Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Teamadmin access for UCCX7 Still Curious to know: Are you able to access the webpage when you follow this path, Start/Programs/Cisco/Desktop/Admin and then Desktop Administrator/Side A ? - Gurpreet On Fri, Mar 23, 2012 at 10:50 PM, Justin McIntyre justin.mcint...@blackbox.commailto:justin.mcint...@blackbox.com wrote: Ok so I was finally able to try the steps that you laid out for me. I first tried to do the steps starting at 7 and this did not work. After that I started at step 1 with the environment variable steps and deleting the teamadmin files. This did not work either. I keep getting this error... HTTP Status 404 - /login.jsp type Status report message /login.jsp description The requested resource (/login.jsp) is not available. Thanks, Justin McIntyre 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. No virus found in this message. Checked by AVG - www.avg.comhttp://www.avg.com Version: 2012.0.1913 / Virus Database: 2114/4890 - Release Date: 03/23/12 inline: image001.gifinline: image002.gifinline: image003.gif___ 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] Teamadmin access for UCCX7
Indeed they arethank you again! Thanks, Justin McIntyre From: Gurpreet Singh Kukreja [mailto:tycoononway1...@gmail.com] Sent: Saturday, March 24, 2012 9:23 AM To: Justin McIntyre Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Teamadmin access for UCCX7 Worked Offline with Justin. everything is working well now. Regards Gurpreet On Sat, Mar 24, 2012 at 8:55 AM, Justin McIntyre justin.mcint...@blackbox.commailto:justin.mcint...@blackbox.com wrote: I'm not sure what Desktop Administrator/Side A is but I can access the Admin utility through programsCiscodesktopadmin. Thanks, Justin McIntyre Engineer Mutual Telecom Services Inc. a wholly-owned subsidiary of Black Box Corp. COMM: (434)-946-1562tel:%28434%29-946-1562 DSN: (312)-237-1562tel:%28312%29-237-1562 CELL: (540)-312-9391tel:%28540%29-312-9391 FAX: (434)-946-1510tel:%28434%29-946-1510 Error! Filename not specified.Error! Filename not specified.Error! Filename not specified. Please note new e-mail address justin.mcint...@blackbox.commailto:justin.mcint...@blackbox.com From: Gurpreet Singh Kukreja [mailto:tycoononway1...@gmail.commailto:tycoononway1...@gmail.com] Sent: Saturday, March 24, 2012 8:51 AM To: Justin McIntyre Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Teamadmin access for UCCX7 Still Curious to know: Are you able to access the webpage when you follow this path, Start/Programs/Cisco/Desktop/Admin and then Desktop Administrator/Side A ? - Gurpreet On Fri, Mar 23, 2012 at 10:50 PM, Justin McIntyre justin.mcint...@blackbox.commailto:justin.mcint...@blackbox.com wrote: Ok so I was finally able to try the steps that you laid out for me. I first tried to do the steps starting at 7 and this did not work. After that I started at step 1 with the environment variable steps and deleting the teamadmin files. This did not work either. I keep getting this error... HTTP Status 404 - /login.jsp type Status report message /login.jsp description The requested resource (/login.jsp) is not available. Thanks, Justin McIntyre 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. No virus found in this message. Checked by AVG - www.avg.comhttp://www.avg.com Version: 2012.0.1913 / Virus Database: 2114/4890 - Release Date: 03/23/12 No virus found in this message. Checked by AVG - www.avg.comhttp://www.avg.com Version: 2012.0.1913 / Virus Database: 2114/4891 - Release Date: 03/24/12 ___ 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 73, Issue 94
I have the same exact problem. I am in Florida and my lab is in Virginia. I have a VPN setup between an 1861 and a 2821(Virginia). Sccp phones function just fine. What kind of service are you getting as your remote location. I am on a install and my hotel only has wireless. Fortunately I brought a spare laptop with me , so I am sharing my wifi connection through my lan connection to my 1861. I belive that this has something to do with it. I also know that there is a short list of supported devices that are to be used for full functionality when building VPN connections like this and my 1861 is not on the list. I'm not sure if any of this was helpful to you as I do not have a solid reason why or the solution to the problem. But yes I had the exact same issues with my sip phones when connected remotely. You might also see that sometimes when you update a phone it loses it's lines but the phone still shows registered. If so delete the main line of the phone from th e system and then rebuild it. Maybe someone here can shed some more light as to why the SIP phones do not stay registered or immediately restart when you try to make an outbound call. I would credit it to port forwarding but I know that the traffic is encrypted by the time it heads out of the outbound interface on the route destined for the VPN server. Also I am using EZVPN configuration in my case with auto connect settings. Hope this helped some. Thanks, Justin McIntyre -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Saturday, March 24, 2012 12:00 PM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 73, Issue 94 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 Firmware Loads (Bryan Byrne) -- Message: 1 Date: Sat, 24 Mar 2012 11:05:33 -0400 From: Bryan Byrne ccie.25...@gmail.com To: ccie_voice voice ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Phone Firmware Loads Message-ID: 381df931-7b88-48f7-b770-6b5025721...@gmail.com Content-Type: text/plain; charset=us-ascii I'm having an odd problem with my home lab. The phone I'm using for HQP2 will have issues sending or receiving calls. The phone will properly register to UCM and it gets dial tone when the phone goes off hook but if I try to dial a number nothing happens. Calling from HQP1 or BR1P2 I get a fast busy. After a minute or two the phone will re-register. If I delete and re-add the phone the problem still happens. The only way I can get the phone back to an operational state is to flip it over to SCCP and then back to SIP. The process of flipping around the phone takes roughly 45 minutes since UCM is in the office and the phones are at my house. Has anyone seen a similar problem? I've duplicated the problem on a 7965 and a 7970. My lab is coming up in 6 weeks and I don't really want to spend any more time troubleshooting. I've got a couple of options 1) Run HQP2 as an SCCP phone. I might might cause some problems with WB 2 labs 2) Install a local CME router to speed up flipping the phones but it's still inconvenient to have to stop in the middle of studying and screw around with phones. Any suggestions? -Bryan -- ___ CCIE_Voice mailing list CCIE_Voice@onlinestudylist.com http://onlinestudylist.com/mailman/listinfo/ccie_voice End of CCIE_Voice Digest, Vol 73, Issue 94 ** - No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.1913 / Virus Database: 2114/4891 - Release Date: 03/24/12 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
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 73, Issue 94
The funny things is the SIP load I am using only acts up when I take it on the road with me, when in the lab no issues at all... Thanks, Justin McIntyre Engineer Mutual Telecom Services Inc. a wholly-owned subsidiary of Black Box Corp. COMM: (434)-946-1562 DSN: (312)-237-1562 CELL: (540)-312-9391 FAX: (434)-946-1510 Please note new e-mail address justin.mcint...@blackbox.com -Original Message- From: Bryan Byrne [mailto:ccie.25...@gmail.com] Sent: Saturday, March 24, 2012 4:50 PM To: Bill Cc: Justin McIntyre; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 73, Issue 94 My setup is just like Justin's. I've got an 1861 with an EzVPN tunnel back into my lab infrastructure. I can easily stand up CME with the right phone firmware to quickly flip the phone. If I get some time on Monday I'm going to do a couple of bug searches to see if there is a problem with the SIP load I'm running. -Bryan On Mar 24, 2012, at 4:29 PM, Bill wrote: adding a local alternate tftp server should solve part of your problem. This way you will not have to pull the sccp or sip firmware over the wan. It is pretty easy to set up on a pc and use to update phone firmware. It might be good to isolate why you have this issue or is it a questionable phone or load? Bill On Mar 24, 2012, at 11:23 AM, Justin McIntyre justin.mcint...@blackbox.com wrote: I have the same exact problem. I am in Florida and my lab is in Virginia. I have a VPN setup between an 1861 and a 2821(Virginia). Sccp phones function just fine. What kind of service are you getting as your remote location. I am on a install and my hotel only has wireless. Fortunately I brought a spare laptop with me , so I am sharing my wifi connection through my lan connection to my 1861. I belive that this has something to do with it. I also know that there is a short list of supported devices that are to be used for full functionality when building VPN connections like this and my 1861 is not on the list. I'm not sure if any of this was helpful to you as I do not have a solid reason why or the solution to the problem. But yes I had the exact same issues with my sip phones when connected remotely. You might also see that sometimes when you update a phone it loses it's lines but the phone still shows registered. If so delete the main line of the phone from th e system and then rebuild it. Maybe someone here can shed some more light as to why the SIP phones do not stay registered or immediately restart when you try to make an outbound call. I would credit it to port forwarding but I know that the traffic is encrypted by the time it heads out of the outbound interface on the route destined for the VPN server. Also I am using EZVPN configuration in my case with auto connect settings. Hope this helped some. Thanks, Justin McIntyre -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Saturday, March 24, 2012 12:00 PM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 73, Issue 94 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 Firmware Loads (Bryan Byrne) -- Message: 1 Date: Sat, 24 Mar 2012 11:05:33 -0400 From: Bryan Byrne ccie.25...@gmail.com To: ccie_voice voice ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Phone Firmware Loads Message-ID: 381df931-7b88-48f7-b770-6b5025721...@gmail.com Content-Type: text/plain; charset=us-ascii I'm having an odd problem with my home lab. The phone I'm using for HQP2 will have issues sending or receiving calls. The phone will properly register to UCM and it gets dial tone when the phone goes off hook but if I try to dial a number nothing happens. Calling from HQP1 or BR1P2 I get a fast busy. After a minute or two the phone will re-register. If I delete and re-add the phone the problem still happens. The only way I can get the phone back to an operational state is to flip it over to SCCP and then back to SIP. The process of flipping around the phone takes roughly 45 minutes since UCM is in the office and the phones are at my house. Has anyone seen a similar problem? I've duplicated the problem on a 7965 and a 7970. My lab
Re: [OSL | CCIE_Voice] Teamadmin access for UCCX7
Ok so I was finally able to try the steps that you laid out for me. I first tried to do the steps starting at 7 and this did not work. After that I started at step 1 with the environment variable steps and deleting the teamadmin files. This did not work either. I keep getting this error... HTTP Status 404 - /login.jsp type Status report message /login.jsp description The requested resource (/login.jsp) is not available. Thanks, Justin McIntyre 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] Teamadmin access for UCCX7
The problem I am having is accessing the page in general. I am getting a http error 500 when I try to go to http://10.10.210.5:6293/teamadmin/login.cda Thanks, Justin McIntyre Engineer Mutual Telecom Services Inc. a wholly-owned subsidiary of Black Box Corp. COMM: (434)-946-1562 DSN: (312)-237-1562 CELL: (540)-312-9391 FAX: (434)-946-1510 [cid:image001.gif@01CD04F9.C0D53B20][cid:image002.gif@01CD04F9.C0D53B20][cid:image003.gif@01CD04F9.C0D53B20] Please note new e-mail address justin.mcint...@blackbox.commailto:justin.mcint...@blackbox.com From: Gurpreet Singh Kukreja [mailto:tycoononway1...@gmail.com] Sent: Sunday, March 18, 2012 6:47 AM To: Justin McIntyre Subject: Re: [OSL | CCIE_Voice] Teamadmin access for UCCX7 Hey Justin, Not sure what we're trying to achieve here. The Access to all teams in system params for uccx is for Supervisor login into the appadmin page for them to be able to access System, RMCM and Help tabs. If i can recall correctly, the Admin has the rights to access CDA and not the supervisor. If you're version 7.0 for CCX, then the default username is admin and the default password is null (i.e., leave it blank and click login) and you should be able to access the CDA Page. Please let us know which way we're heading? Regards Gurpreet On Sat, Mar 17, 2012 at 7:15 PM, Justin McIntyre justin.mcint...@blackbox.commailto:justin.mcint...@blackbox.com wrote: Hi guys/gals, I'm having issues accessing http://10.10.210.5:6293/teamadmin/login.cda The Cisco agent login. I have enabled access to all teams in system parameters and made one of the users a Supervisor. However I am getting a login.jsp is unavailable page Http error 500. Any one had any experience with this issue that they can help with? Thanks in advance... Thanks, Justin McIntyre 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.comhttp://www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.PlatinumPlacement.com No virus found in this message. Checked by AVG - www.avg.comhttp://www.avg.com Version: 2012.0.1913 / Virus Database: 2114/4876 - Release Date: 03/17/12 inline: image001.gifinline: image002.gifinline: image003.gif___ 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] IPEXPERT.com
I guess while the ipexpert system gets updated we're going to get some downtime eh? Can't get to my lab pdfs until website used for authentication comes back up Bore Thanks, Justin McIntyre 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
[OSL | CCIE_Voice] Teamadmin access for UCCX7
Hi guys/gals, I'm having issues accessing http://10.10.210.5:6293/teamadmin/login.cda The Cisco agent login. I have enabled access to all teams in system parameters and made one of the users a Supervisor. However I am getting a login.jsp is unavailable page Http error 500. Any one had any experience with this issue that they can help with? Thanks in advance... Thanks, Justin McIntyre 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
[OSL | CCIE_Voice] [OSL | CCIE_VOICE] SRST and MWI unsolicited notify.
I am wondering if anyone one can give me some hints on where to troubleshoot swi updates not working with unsolicited notify. I have my ccn subsystem output and my running config input below. My issue is that I do not see any Sip Notify messages being sent out. SHOW CCN SUB SIP se-10-10-115-2# show ccn subsystem sip SIP Gateway:10.10.115.1 SIP Port Number:5060 DTMF Relay: sip-notify,sub-notify MWI Notification: unsolicited MWI Envelope Info: disabled Transfer Mode: bye-also SIP RFC Compliance: Pre-RFC3261 Running config Current configuration : 9319 bytes ! ! Last configuration change at 03:20:35 GMT Sun Mar 18 2012 by justin ! NVRAM config last updated at 03:20:30 GMT Sun Mar 18 2012 by justin ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname SiteC-RTR ! boot-start-marker boot-end-marker ! logging message-counter syslog ! no aaa new-model memory-size iomem 20 clock timezone GMT 0 network-clock-participate wic 3 network-clock-select 1 E1 0/3/0 dot11 syslog ip source-route ! ! ip dhcp excluded-address 10.10.202.1 10.10.202.119 ip dhcp excluded-address 10.10.202.130 10.10.202.254 ! ip dhcp pool SiteC-Static host 10.10.202.130 255.255.255.0 client-identifier 0100.1930.5d0b.d7 default-router 10.10.202.1 option 150 ip 10.10.210.11 10.10.210.10 ! ip dhcp pool SiteC-PHONES network 10.10.202.0 255.255.255.0 default-router 10.10.202.1 option 150 ip 10.10.210.11 10.10.210.10 ! ! ip cef ip domain name ipexpert.com no ipv6 cef ! multilink bundle-name authenticated ! ! ! ! isdn switch-type primary-ni ! ! ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip sip ! ! ! voice class codec 1 codec preference 1 g729r8 codec preference 2 g711ulaw ! ! ! ! voice class h323 1 h225 timeout tcp establish 3 ! ! ! ! voice class custom-cptone exit dualtone conference cadence 400 ! voice class custom-cptone entry dualtone conference cadence 200 ! ! ! ! ! ! ! voice translation-rule 1 rule 1 /.+\(\)$/ /\1/ ! voice translation-rule 2 rule 1 /^4...$/ /7796\0/ ! voice translation-rule 900 rule 1 /^4...$/ /+144207796\0/ ! ! voice translation-profile 4digitDNIS translate called 1 ! voice translation-profile 8digitANI translate calling 2 ! voice translation-profile e164ANI translate calling 900 ! ! voice-card 0 dsp services dspfarm ! ! crypto pki trustpoint TP-self-signed-1655997933 enrollment selfsigned subject-name cn=IOS-Self-Signed-Certificate-1655997933 revocation-check none rsakeypair TP-self-signed-1655997933 ! ! crypto pki certificate chain TP-self-signed-1655997933 certificate self-signed 01 3082024C 308201B5 A0030201 02020101 300D0609 2A864886 F70D0101 04050030 31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274 69666963 6174652D 31363535 39393739 301E 170D3132 30323131 31343135 30385A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649 4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D31 36353539 39373933 3330819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281 8100E6E4 0307318B 5E2C94CB 7E2A83CF 6F99AE89 10D93A2F 38BDEB71 95C5695E 4BEA4075 6AE144A6 961F0630 CCECF324 EDB7E128 64BA6A7F 289758F4 8C5268BF C36E7746 40F8CDEE 8D5EE734 0BADF088 8B1B933F BFC9CD9C B25CC8C7 D68AFDEC FC8AC19F 6200D364 7F82FD03 7B43C688 DF02DF00 31F09D24 A21421D8 26CA303C ACDF0203 010001A3 74307230 0F060355 1D130101 FF040530 030101FF 301F0603 551D1104 18301682 14425232 2D525452 2E697065 78706572 742E636F 6D301F06 03551D23 04183016 80142071 A4496B36 760E3BB9 7BA7ECB2 3441D434 EA54301D 0603551D 0E041604 142071A4 496B3676 0E3BB97B A7ECB234 41D434EA 54300D06 092A8648 86F70D01 01040500 03818100 342F96C6 47F5E13E 1EB508A2 6A614A3F 9C975E35 B6690F3A 74E75E4D E88F802B 6A09E40D 3E86128D BDFD34EC D2C0FF33 E3DDB0B8 495F5600 A1921326 11E4851E DED6D532 C2B597B9 1755F18E 8A71C86B A6D3D77A E10E2868 6D8B6B13 20988D7D 4ABF185D 7332B029 F418C6A8 02408832 E818FE78 3E8DD234 71E50FB7 C39A5141 quit ! ! username justin privilege 15 secret 5 $1$WDlv$sbhmjA4R4UJnOVIG3A1iN/ archive log config hidekeys ! ! ! ! ! controller E1 0/3/0 channel-group 4 timeslots 4-15,17-31 pri-group timeslots 1-3,16 ! ip ssh version 2 ! ! ! ! interface Loopback0 ip address 10.10.110.3 255.255.255.255 ip ospf network point-to-point h323-gateway voip bind srcaddr 10.10.110.3 ! interface Loopback1 ip address 10.10.115.1 255.255.255.0 ip ospf network point-to-point ! interface FastEthernet0/0 no ip address duplex auto speed auto ! interface FastEthernet0/0.301 encapsulation dot1Q 301 native ip address 10.10.102.1 255.255.255.0 ! interface FastEthernet0/0.302 encapsulation dot1Q 302 ip address
Re: [OSL | CCIE_Voice] CME/CUE DTMF relay from UCM(sip trunk) to CME/CUE (Juan Lopez)
Hello, I just wanted to update that the SIP-Notify did not work when set at the UCM SIP trunk, the BR2 CUE Dial-peer and also within CUE configuration. At this point it only seems to be working when using the MTP to terminate between UCM SIP trunk and CME dial-peer. Thanks, Justin McIntyre Engineer Mutual Telecom Services Inc. a wholly-owned subsidiary of Black Box Corp. COMM: (434)-946-1562 DSN: (312)-237-1562 CELL: (540)-312-9391 FAX: (434)-946-1510 Please note new e-mail address justin.mcint...@blackbox.com -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Sunday, March 04, 2012 1:23 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 73, Issue 8 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: service-policy on trunk ports (Vik Malhi) 2. Re: CME/CUE DTMF relay from UCM(sip trunk) to CME/CUE (Juan Lopez) 3. Re: CME/CUE DTMF relay from UCM(sip trunk) to CME/CUE (Juan Lopez) -- Message: 1 Date: Sat, 3 Mar 2012 19:00:34 -0800 From: Vik Malhi vma...@ipexpert.com To: Ken Wyan kew...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] service-policy on trunk ports Message-ID: cada9c18-6add-4058-b744-194805b8d...@ipexpert.com Content-Type: text/plain; charset=windows-1252 This is and has been for a long time been a limitation on the 3750- the show policy-map command doesn't work:-( Vik Malhi ? CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Mar 3, 2012, at 12:59 PM, Ken Wyan wrote: I have following scenario (Tested in Proctorlabs Rack). HQ Switch Fa1/0/1 (trunk port) ---connect to-- HQ Router Fa0/0 (with sub-interfaces) I want to apply a service-policy to mgcp packets going through this link. I configured access-list , class-map , policy-map applied to switch interface. But I can't see any mgcp packets matching HQ-3750#show policy-map interface fastEthernet 1/0/1 FastEthernet1/0/1 Service-policy input: mgcp Class-map: mgcp (match-all) 0 packets, 0 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: access-group 100 Class-map: class-default (match-any) 0 packets, 0 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: any 0 packets, 0 bytes 5 minute rate 0 bps interface FastEthernet1/0/1 switchport trunk encapsulation dot1q switchport trunk native vlan 10 switchport mode trunk speed 100 duplex full mls qos trust dscp service-policy input mgcp Now same thing I configured on HQ Router ( Fa0/0 interface) , then I can see packets are matching with service policy. What can be the reason? (Switch accepts service-policy in input direction only , hence I applied service-policy in output direction on Router port) Can this be a limitation for trunk (multi-vlan) ports on switches ? Ken ___ 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/20120303/1ed256a1/attachment-0001.html -- Message: 2 Date: Sun, 4 Mar 2012 06:34:42 +0100 From: Juan Lopez lopez.hernandez.j...@gmail.com To: Justin McIntyre justin.mcint...@blackbox.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME/CUE DTMF relay from UCM(sip trunk) to CME/CUE Message-ID: canpj6cze1nfeesk2ou6pgudzgo9cbyd5nuffqboxkdzdptk...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi Justin, from reading the mail it looks like on the SIP dialpeers on the BR2, you use the rtp-nte (inband) dtmf-relay method? can you try and let us know: 1) use SIP-NOTIFY on both SIP dialpeers at BR2? (not sure if UCM supports this - in SRND it states a UCM SIP trunk uses RTP-NTE or possibly SIP-KPML) if 1 does not work: 2) use sip notify on CUE dialpeer and sip-kpml on sip dialpeer to UCM. Not sure here whether the CUBE at branch 2 supports notify - kpml dtmf interworking
[OSL | CCIE_Voice] Cisco Unified Presence Server Version usage
Good Day everyone, My Question today is in regards to the Presence Server interoperability with UCM 7.0.1. I have CUPS 7.0.0 and it turns out that there is a good difference in function between CUPS 7.0.1 and CUPS 7.0.0. I have the 8.0 NFR kit and did some testing with the integration of CUCM 7.0.1 and CUPS 8.0.2 however I am seeing some difficulties in getting the system to update user information across from UCM towards CUPS. I.E. I am having to reset CUPS to get it to resync information that I have updated on the CUCM side of the integration. I'm sure a lot of that has to do with my experience and knowledge level of the software release upon which I am working. What I am looking for is if anyone has used CUPS 8.0 in place of 7.0.1 for their CCIE Voice studies. I have not been able to find the NFR software for 7.0.1. It seems that it has been discontinued for sale. I was able to come up with all the other required software other than CUPS which as I stated I have 7.0.0.9000xxx. Thank you in advanced for the help. Thanks, Justin McIntyre 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
[OSL | CCIE_Voice] CUPS 7.0.1
Good Day everyone, My Question today is in regards to the Presence Server interoperability with UCM 7.0.1. I have CUPS 7.0.0 and it turns out that there is a good difference in function between CUPS 7.0.1 and CUPS 7.0.0. I have the 8.0 NFR kit and did some testing with the integration of CUCM 7.0.1 and CUPS 8.0.2 however I am seeing some difficulties in getting the system to update user information across from UCM towards CUPS. I.E. I am having to reset CUPS to get it to resync information that I have updated on the CUCM side of the integration. I'm sure a lot of that has to do with my experience and knowledge level of the software release upon which I am working. What I am looking for is if anyone has used CUPS 8.0 in place of 7.0.1 for their CCIE Voice studies. I have not been able to find the NFR software for 7.0.1. It seems that it has been discontinued for sale. I was able to come up with all the other required software other than CUPS which as I stated I have 7.0.0.9000xxx. Thank you in advanced for the help. Thanks, Justin 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] MGCP Registration
I noticed two things... 1. It seems that the ccm-manager config server is there but not ccm-manager config Not sure if that's absolutely needed or not but I have always put it in if I am going to let the CM configure the gateway. 2. I see that control is bound to a sub-interface. I'm pretty sure that there is a bug with this. I would try to do a no isdn bind-l3 ccm-manager and then isdn bind-l3 ccm-manager on the voice serial interface and see if it brings the layer 3 status up to Multi-Frame Established Thanks, Justin McIntyre Engineer Mutual Telecom Services Inc. a wholly-owned subsidiary of Black Box Corp. COMM: (434)-946-1562 DSN: (312)-237-1562 CELL: (540)-312-9391 FAX: (434)-946-1510 Please note new e-mail address justin.mcint...@blackbox.com -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Monday, January 16, 2012 9:02 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 71, Issue 61 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: MGCP Registration (George Goglidze) -- Message: 1 Date: Mon, 16 Jan 2012 15:01:33 +0100 From: George Goglidze gogli...@gmail.com To: mercy forall mercy_for_...@hotmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MGCP Registration Message-ID: ca+dn5iosl6uyfzpbdaj6qsbwwo3nuniscg6mh8zdukhjaps...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi Mercy, Did you configure this via tftp or manually? I'm asking because you've got this: ccm-manager config server x.x.x.x Can you do: *debug ccm-manager config-download all *then do * no ccm-manager config ccm-manager config* And see if you are being able to download the config. paste the output here. As well you can attach the relevant pages of CCM configuration. Cheers, * * On Mon, Jan 16, 2012 at 2:35 PM, mercy forall mercy_for_...@hotmail.comwrote: Hi, thanks for your support and good link\ this is my GW configuration , also it is connected to other cisco GW as PSTN GW through E1 cross cable sh run : Current configuration : 15381 bytes ! ! version 15.0 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname ! boot-start-marker boot-end-marker ! logging buffered 51200 warnings no aaa new-model network-clock-participate wic 2 ! dot11 syslog ip source-route ! ip cef ! ! no ipv6 cef multilink bundle-name authenticated ! ! ! isdn switch-type primary-qsig ! voice-card 0 ! ! voice rtp send-recv ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip redirect ip2ip h323 sip header-passing no call service stop ! voice class codec 1 codec preference 1 g711ulaw ! voice class custom-cptone dualtone disconnect frequency 425 cadence 250 250 ! ! ! ! http client cache memory pool 15000 http client cache memory file 500 http client connection timeout 60 http client connection idle timeout 10 http client response timeout 30 mrcp client timeout connect 10 mrcp client timeout message 10 mrcp client rtpsetup enable vxml tree memory 500 vxml audioerror vxml version 2.0 ! crypto pki trustpoint TP-self-signed-3307538538 enrollment selfsigned subject-name cn=IOS-Self-Signed-Certificate-3307538538 revocation-check none rsakeypair TP-self-signed-3307538538 ! controller E1 0/2/0 pri-group timeslots 1-4,16 service mgcp ! interface GigabitEthernet0/0 no ip address duplex auto speed auto media-type rj45 ! interface GigabitEthernet0/0.1 encapsulation dot1Q ip address X.X.X.X 255.255.255.0 ! ! interface Serial0/2/0:15 no ip address encapsulation hdlc isdn switch-type primary-qsig isdn incoming-voice voice isdn bind-l3 ccm-manager no cdp enable ! ip forward-protocol nd ! ! ip http server ip http access-class 23 ip http authentication local ip http secure-server ip http timeout-policy idle 60 life 86400 requests 1 ip route 0.0.0.0 0.0.0.0 X.X.X.X ! ! ! control-plane ! call threshold global cpu-5sec low 70 high 85 ! voice-port 0/2/0:15 ! voice-port 0/3/0 ! voice-port 0/3/1 ! ccm-manager switchback immediate ccm-manager fallback-mgcp ccm-manager redundant
[OSL | CCIE_Voice] Gatekeeper Codec control
Hello everyone. My question today is concerning controlling which codecs are used when utilizing RAS signaling via the Gatekeeper. I understand that I can control my codec inbound (at the BR2 CME) site via a inbound dial-peer that only utilizes g729r8. I also understand that using Transcoding at the same location will allow me to talk to a locally attached SIP phone (at CME site) that is configured to use g711ulaw only. However I am unclear as to how to program the CUCM controlled devices what codec to use when sourcing calls to the BR2 site via gatekeeper. If I am sourcing calls from CUCM across a gatekeeper trunk that has been configured to be in the HQ device pool which is associated with the HQ region which uses g711 intra-cluster... then shouldn't I be sourcing packets from CUCM to BR2 as g711ulaw? Any additional thoughts or clarification would be greatly appreciated. Thanks, Justin McIntyre Engineer Mutual Telecom Services Inc. a wholly-owned subsidiary of Black Box Corp. COMM: (434)-946-1562 DSN: (312)-237-1562 CELL: (540)-312-9391 FAX: (434)-946-1510 [cid:image001.gif@01CCCFA6.E516CFE0][cid:image002.gif@01CCCFA6.E516CFE0][cid:image003.gif@01CCCFA6.E516CFE0] Please note new e-mail address justin.mcint...@blackbox.commailto:alex.heve...@blackbox.com 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. inline: image001.gifinline: image002.gifinline: image003.gif___ 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 Enabled Secret (Justin McIntyre)
All is resolved. Thanks to those who helped out so quickly. I was able to send a break command after the image decompressed and then reset to factory defaults. I love getting to learn stuff like this. From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com [ccie_voice-requ...@onlinestudylist.com] Sent: Saturday, December 31, 2011 1:08 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 70, Issue 161 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 70, Issue 157 (Sadaseeven Saminaden) 2. Telephony-Service vs call-manager-fallback (Randall Crumm) -- Message: 1 Date: Sat, 31 Dec 2011 07:04:43 +0400 From: Sadaseeven Saminaden sadaseev...@emenetworks.com To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 70, Issue 157 Message-ID: cajnyshxze9ax_32rzg2dietjrez2wbx3hmvsx5axhvgq3w3...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 hi randall. Do you have a local username defined on the router? Also, did you try logging in via the service module's console? On Dec 31, 2011 3:56 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. E1 configuration (Eliot Ngwa) (michael.se...@compucom.com) 2. PSTN-WAN Enabled Secret (Justin McIntyre) 3. CUE wizard authentication issue (Randall Crumm) 4. Re: CUE wizard authentication issue (linuxboss.9) 5. Re: CUE wizard authentication issue (Randall Crumm) -- Message: 1 Date: Fri, 30 Dec 2011 13:40:32 -0500 From: michael.se...@compucom.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] E1 configuration (Eliot Ngwa) Message-ID: 426D14439C8C604B90E332DD4696917301BD604092@SP049EXC32.compucom.local Content-Type: text/plain; charset=us-ascii I read in the thread below I am using simple crossover cable (Ethernet crossover). This cable will not work. You need a T1 cross over cable: http://www.google.com/search?q=t1+crossover+cablehl=enprmd=imvnstbm=ischtbo=usource=univsa=Xei=3gP-Tp2AIpS5twf-ufXPBgsqi=2ved=0CF4QsAQbiw=1072bih=804 !!! If your using an Ethernet cross over it won't work need T1 cross over 1--4, 2--5, 4--1, 5--2 http://www.ebay.com/itm/T1-Crossover-cable-3FT-/160570999135?pt=LH_DefaultDomain_0hash=item2562c7015f Usually you can pick one up at local computer store or ebay real cheap depending on the length. Hope this helps. Michael Sears -- Message: 2 Date: Fri, 30 Dec 2011 17:12:06 -0500 From: Justin McIntyre justin.mcint...@blackbox.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] PSTN-WAN Enabled Secret Message-ID: 0a566999c353b042841473fa64165cd62e7390d...@exchcluster.corp.bbns.com Content-Type: text/plain; charset=iso-8859-1 Hello everyone, I am new and i am stumbling right out of the gate. I have searched on this and have been unable to find the exact answer i'm looking for. I have uploaded the PSTN-WAN configs to my home lab and of course i forgot to take our the enabled secret line as well as the no service password-recovery command. I am in need of either the password or a way to restore my device to factory defaults. I am in the interim still searching and hounding for a solution but any assistance would be greatly appreciated. The device is a(n) Cisco 2801. Thank you in advanced for the assistance. This email and any files transmitted with it are confidential and are intended for the sole use of the individual to whom
[OSL | CCIE_Voice] PSTN-WAN Enabled Secret
Hello everyone, I am new and i am stumbling right out of the gate. I have searched on this and have been unable to find the exact answer i'm looking for. I have uploaded the PSTN-WAN configs to my home lab and of course i forgot to take our the enabled secret line as well as the no service password-recovery command. I am in need of either the password or a way to restore my device to factory defaults. I am in the interim still searching and hounding for a solution but any assistance would be greatly appreciated. The device is a(n) Cisco 2801. Thank you in advanced for the assistance. 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
