Re: [OSL | CCIE_Voice] IP PIM
Dense mode is a push method of doing multicast and sparse dense is a pull method of doing multicast. On Tue, Mar 3, 2009 at 12:56 PM, hasan khalife hasan_khal...@hotmail.comwrote: WHAT IS THE DIFFERENEC BTW IP PIM DENSE-MODE IP PIM SPARSE-DENSE-MODE -- See all the ways you can stay connected to friends and familyhttp://www.microsoft.com/windows/windowslive/default.aspx
Re: [OSL | CCIE_Voice] TFTP Server best practices
It should be spelled out, but if it isn't, generally best practice is Publisher runs TFTP server. Thanks, Ryan Trauernicht CCIE Voice #23497 On Tue, Feb 24, 2009 at 12:20 AM, Chris Parker cpar...@cparker.us wrote: What configuration should we use for the TFTP servers if it is not spelled out explicitly in our lab? A single TFTP on the PUB? Or on both SUB and PUB? I guess the safest bet is just to have it on on both?
Re: [OSL | CCIE_Voice] Antw: ATA186 dot1q ot not?
Robert is correct. the ATA 186 you should configure as an access port and not dot1q. The ATA188 (which have a PC port) you would configure as a dot1q port, but they are not on the lab. Thank, Ryan Trauernicht CCIE Voice #23497 On Sun, Feb 22, 2009 at 8:53 AM, Robert Schuknecht rschukne...@gmx.dewrote: Hi Chris, i configure the switchport for the ATA always as access-port. I would only configure the switchport for dot1q if i am asked to explicitly. /Robert Chris Parkercpar...@cparker.us schrieb am Sonntag, 22. Februar 2009 um 14:48 in Nachricht b74096f513403f0e877b6a80074dcdb5: Should the port connecting to the ATA186 be dot1q or not? Since the 186 doesn't have an ethernet port for a PC I guess its optional?
Re: [OSL | CCIE_Voice] Unity Voicemail Port Configuration
I would exclude the MWI port in the line group. If it is not answering calls (you have unchecked that function on the unity side) there is no point in putting it in the line group. Thanks, Ryan Trauernicht CCIE Voice #23497 On Sun, Feb 22, 2009 at 6:05 AM, Robert Schuknecht rschukne...@gmx.dewrote: Hi List, lets assume we are required to configure Unity with 3 Ports + 1 MWI Port. Normally all Voicemail Ports are in a Line Group. Would you leave the MWI Voicemail Port also in the Line Group or would you exclude the MWI Voicemail Port from the Line Group? /Robert
Re: [OSL | CCIE_Voice] B-Channel Maintenance not working properly
When you try this again what does Perfmon tell you the service of the channels? 0, 1, 2, 3? On Sun, Feb 22, 2009 at 11:07 AM, Robert Schuknecht rschukne...@gmx.dewrote: Hi List, during my last Remote-Rack Sessions i noticed that the B-Channel Maintenance Status Parameter is not workking properly. Always when i configured it and restartet the CCM Srevice and the Gateway itself, it is working for some calls. And suddenly the Gateway is trying to call out over the not available B-Channels I configured the Maintenance Status the following ways: 1) S0/ds...@sda000332333241=0001 2) S0/ds...@sda000332333241 = 0001 But both of them did not work. What am i doing wrong here? /Robert
Re: [OSL | CCIE_Voice] Block incoming calls on H323 GW in SRST
You want to use corlist. Your incoming dial-peer for all calls (regular mode or SRST mode) you apply a corlist incoming. On the call-manager-fallback you apply a corlist of the DN you want to block in the outgoing direction. Thanks, Ryan Trauernicht CCIE Voice #23497 On Wed, Feb 18, 2009 at 10:17 PM, Balamurugan Singaram mmailb...@yahoo.comwrote: working with COR list is the best solution for MGCP gateway I think so..when I try the same in h.323 gaeway the call is blocked all the time. Could please some one share the best solution for blocking calls in h.323 gateway to. Thanks --- On *Thu, 19/2/09, Jose Gregorio Linero (jlinero) jlin...@cisco.com*wrote: From: Jose Gregorio Linero (jlinero) jlin...@cisco.com Subject: Re: [OSL | CCIE_Voice] Block incoming calls on H323 GW in SRST To: DIEGO FERNANDO MACIAS SANCHEZ dmac...@javeriana.edu.co, ccie_voice@onlinestudylist.com Date: Thursday, 19 February, 2009, 4:15 AM Hi Try to use cor list. Regards, Jose -- *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *DIEGO FERNANDO MACIAS SANCHEZ *Sent:* Miércoles, Febrero 18, 2009 5:42 PM *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] Block incoming calls on H323 GW in SRST Hello all Does anybody know the way to block incoming calls from PSTN to an especific extension connected to an H323 GW. If i apply a blocking translation rule, this will be active in normal operation also. Regards DM AVISO LEGAL: El presente correo electronico no representa la opinion o el consentimiento oficial de la PONTIFICIA UNIVERSIDAD JAVERIANA. Este mensaje es confidencial y puede contener informacion privilegiada la cual no puede ser usada ni divulgada a personas distintas de su destinatario. Esta prohibida la retencion, grabacion, utilizacion, aprovechamiento o divulgacion con cualquier proposito. Si por error recibe este mensaje, por favor destruya su contenido y avise a su remitente. En este aviso legal se omiten intencionalmente las tildes. -- Get your preferred Email name! http://sg.rd.yahoo.com/aa/mail/domainchoice/mail/signature/*http://mail.promotions.yahoo.com/newdomains/aa/ Now you can @ymail.com and @rocketmail.com.
Re: [OSL | CCIE_Voice] SRST VM: TP or VM-Integration ?
The VM-Integration is for FXO ports. Unless your PSTN router will route a pattern like 3125551212#4001 (which is prob not the case in the lab or the real world). If you run into the RDNIS bug (FF) in the lab you will need to find another work around. You can prob use the Translation Pattern or the CTI Route Point workaround. Thanks, Ryan Trauernicht CCIE Voice #23497 On Sat, Feb 14, 2009 at 1:29 PM, Mike Brooks 2xcci...@gmail.com wrote: So which is the preferred method to get VM to work in SRST mode ? The translation pattern method or the vm-integration method ? To me it seems that alot of people are have issues getting the vm-integration method to work. Therefore, currently I only use the translation-pattern method. I have never been able to get the vm-intergration method to work properly. If you are able to get the vm-integration method to work please post your config. Thx, Mike Brooks CCIE# 16027 (RS)
[OSL | CCIE_Voice] CCIE Passed!!!
I wanted to let everyone know that I passed my CCIE Voice yesterday!! Lucky attempt 1... i couldnt believe it! I wanted to thank everyone from this list that helped out in my studying for the last 7 months. I could not have done it without all the brainstorming and creativity that is shared on this list. thank you all again!
Re: [OSL | CCIE_Voice] CCIE Passed!!!
I forgot to post my number... thanks again everyone! Ryan Trauernicht CCIE Voice #23497 On Wed, Feb 11, 2009 at 3:59 PM, Kumar, Narinder narinder.ku...@uxcg.com.au wrote: Well Done Ryan. Congratulations. *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Ryan Trauernicht *Sent:* Thursday, 12 February 2009 8:34 AM *To:* OSL Group *Subject:* [OSL | CCIE_Voice] CCIE Passed!!! I wanted to let everyone know that I passed my CCIE Voice yesterday!! Lucky attempt 1... i couldnt believe it! I wanted to thank everyone from this list that helped out in my studying for the last 7 months. I could not have done it without all the brainstorming and creativity that is shared on this list. thank you all again! -- CONFIDENTIALITY - The information contained in this electronic mail message is confidential and is intended solely for the addressee(s). If you are not an authorised recipient of this message please contact Getronics Australia immediately by reply email and destroy/delete this message from your computer. Any unauthorised form of reproduction of this message, or part thereof, is strictly prohibited. DISCLAIMER - Unless specifically indicated otherwise, the views and opinions expressed in this email are those of the sender and not Getronics Australia. While we endeavour to protect our network from computer viruses, Getronics Australia does not warrant that this email or any attachments are free of viruses or any other defects or errors. It is the duty of the recipient to virus scan and otherwise test any information contained in this email before loading onto any computer system.
Re: [OSL | CCIE_Voice] BACD Refresh
Yup. That will def do it for you. If you have term mon turned on you should see a log come across saying it loaded successfully. Thanks, Ryan Trauernicht On Mon, Feb 9, 2009 at 10:21 PM, Chris Parker cpar...@cparker.us wrote: Is: call application voice load aa call application voice load queue the way to go? Chris Parker wrote: When you make a change to the BACD parameters or prompts after they have loaded, the BACD application needs to be reloaded for the changes to take affect. What is the best way to do this short of reloading the router? Chris
Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
under the Media section. This leads me to believe they want you to use the annunciator. Otherwise wouldn't it be under the Voicemail/Unity section? Regardless I don't think you can do it any other way unless you hairpin the call through Unity to send the call to the annunciator since the VM ports are skinny registrations. Chris Ryan Trauernicht wrote: That is what I thought but I opened a TAC case and they claim you can, but cant figure out how. Thanks, Ryan Trauernicht On Mon, Jan 26, 2009 at 3:21 AM, Juan juan.c...@gmail.com mailto:juan.c...@gmail.com wrote: I remember reading in the SRND that you can only engage the annunciator for SCCP devices if I remember correctly - so not to the PSTN. cheers, Juan On Sun, Jan 25, 2009 at 5:48 PM, Ryan Trauernicht ryanstudyvo...@gmail.com mailto:ryanstudyvo...@gmail.com wrote: Not sure why you are going through all that trouble and not just sending it to unity as a call handler and hang up after message played. I don't know how to play an ANN from a PSTN call, I have engaged TAC and they are still working on it and they can't even figure it out right now. Any ideas? Thanks, Ryan Trauernicht On Sun, Jan 25, 2009 at 3:42 AM, Kapil Atrish kapilatr...@hotmail.com mailto:kapilatr...@hotmail.com wrote: Hi list, Following I did: Create a new MOH Audio Source using AAExtnOutOfService.wav. Prompt available inside Wfavvid folder Create a TP covering all unassigned DNs for example: 11xx, do Called party Xform to 1155 Create a AC Pilot 1155, give any DP say: ANN_PSTN AC Hunt-GroupGive any AC user. No need to login to Attendant Console. Run acconfig.batEnable Queuing Inside DP: ANN_PSTN give User Hold MOH Source as AAExtnOutOfService.wav. Now, whenever you dial any unassigned number withing range 11xx, you'll hear AAExtnOutOfService.wav but the problem is that I am not able to make the PSTN call drop. I tried routing calls to TP inside AC Hunt-GroupAlways Route member is TPTP has Block Pattern --Not working. AAExtnOutOfService.wavkeeps on playing. I tried routing calls to Route-Point (Always Route Member) inside AC Hunt-GroupCTI_RP has Forward all to TPTP has Block Pattern --Not working. AAExtnOutOfService.wavkeeps on playing. I tried routing calls to a registered Phone DN as Always Route MemberForward all to TPTP has Block Pattern --Not working. AAExtnOutOfService.wavkeeps on playing. Can someone help me achieve call drop here without using IPCCX/Unity/TCL? Thanks, Kapil Atrish Rediscover the magic of Windows WIN a Windows Vista laptop Windows mobile phone at www.windowsandme.com http://www.windowsandme.com Try it now! http://www.windowsandme.com __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ Need more space to upload pictures? Get 25 GB online storage with Windows Live SkyDrive! Try it! Discover your phone style WIN a Windows Mobile phone. Your style! Try it now! http://www.whatsmyphonestyle.com __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ Akshay Kumar takes
Re: [OSL | CCIE_Voice] IPMA Assistant App
I have configured it on both servers. Still comes up with the same error. Thanks, Ryan Trauernicht On Sat, Feb 7, 2009 at 6:07 AM, Christian Hennrich christian.hennr...@intact-is.com wrote: hi, have you configured the ipma server on sub and pub. the ipma service parameter need to be configure on both. they are not global. that resolved the problem for me HTH Ryan Trauernicht schrieb: yup. On Fri, Feb 6, 2009 at 8:07 PM, anil batra anil...@yahoo.com mailto: anil...@yahoo.com wrote: Is the IPMA service ON Sub --- On *Sat, 2/7/09, Ryan Trauernicht /ryanstudyvo...@gmail.com mailto:ryanstudyvo...@gmail.com/* wrote: From: Ryan Trauernicht ryanstudyvo...@gmail.com mailto:ryanstudyvo...@gmail.com Subject: [OSL | CCIE_Voice] IPMA Assistant App To: OSL Group ccie_voice@onlinestudylist.com mailto:ccie_voice@onlinestudylist.com Date: Saturday, February 7, 2009, 6:44 AM When the app boot ups it asks for the IPMA server. In the service params I set the Sub first and Pub as backup. If I put in the IP of the Sub it errors out and said it can not find the IPMA server. Is this normal? Thanks, Ryan Trauernicht __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __
Re: [OSL | CCIE_Voice] IPMA Assistant App
I have tried that as well. I reboot both the sub and pub. Issue still continues. On Sat, Feb 7, 2009 at 11:16 AM, basant yadav basant.ya...@gmail.comwrote: Hi Ryan I just had the same issue a few minutes back. While choosing subscriber as server for IPMA console, getting error cannot find server however when selecting Publisher in place, it worked fine. I fixed it by restarting the Tomcat service on subscriber using services.msc followed by rebooting the subscriber. HTH - Basant On Sat, Feb 7, 2009 at 6:09 PM, Ryan Trauernicht ryanstudyvo...@gmail.com wrote: I have configured it on both servers. Still comes up with the same error. Thanks, Ryan Trauernicht On Sat, Feb 7, 2009 at 6:07 AM, Christian Hennrich christian.hennr...@intact-is.com wrote: hi, have you configured the ipma server on sub and pub. the ipma service parameter need to be configure on both. they are not global. that resolved the problem for me HTH Ryan Trauernicht schrieb: yup. On Fri, Feb 6, 2009 at 8:07 PM, anil batra anil...@yahoo.com mailto: anil...@yahoo.com wrote: Is the IPMA service ON Sub --- On *Sat, 2/7/09, Ryan Trauernicht /ryanstudyvo...@gmail.com mailto:ryanstudyvo...@gmail.com/* wrote: From: Ryan Trauernicht ryanstudyvo...@gmail.com mailto:ryanstudyvo...@gmail.com Subject: [OSL | CCIE_Voice] IPMA Assistant App To: OSL Group ccie_voice@onlinestudylist.com mailto:ccie_voice@onlinestudylist.com Date: Saturday, February 7, 2009, 6:44 AM When the app boot ups it asks for the IPMA server. In the service params I set the Sub first and Pub as backup. If I put in the IP of the Sub it errors out and said it can not find the IPMA server. Is this normal? Thanks, Ryan Trauernicht __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __
Re: [OSL | CCIE_Voice] IPMA Assistant App
CTI or IPMA logs? IPMA on the Sub do not really have any logs even after changing it to detailed. On Sat, Feb 7, 2009 at 11:25 AM, basant yadav basant.ya...@gmail.comwrote: In that case, Pls collect and send IPMA console logs. Lets see why its failing to connect to SUB. - Basant On Sat, Feb 7, 2009 at 6:20 PM, Ryan Trauernicht ryanstudyvo...@gmail.com wrote: I have tried that as well. I reboot both the sub and pub. Issue still continues. On Sat, Feb 7, 2009 at 11:16 AM, basant yadav basant.ya...@gmail.comwrote: Hi Ryan I just had the same issue a few minutes back. While choosing subscriber as server for IPMA console, getting error cannot find server however when selecting Publisher in place, it worked fine. I fixed it by restarting the Tomcat service on subscriber using services.msc followed by rebooting the subscriber. HTH - Basant On Sat, Feb 7, 2009 at 6:09 PM, Ryan Trauernicht ryanstudyvo...@gmail.com wrote: I have configured it on both servers. Still comes up with the same error. Thanks, Ryan Trauernicht On Sat, Feb 7, 2009 at 6:07 AM, Christian Hennrich christian.hennr...@intact-is.com wrote: hi, have you configured the ipma server on sub and pub. the ipma service parameter need to be configure on both. they are not global. that resolved the problem for me HTH Ryan Trauernicht schrieb: yup. On Fri, Feb 6, 2009 at 8:07 PM, anil batra anil...@yahoo.commailto: anil...@yahoo.com wrote: Is the IPMA service ON Sub --- On *Sat, 2/7/09, Ryan Trauernicht /ryanstudyvo...@gmail.com mailto:ryanstudyvo...@gmail.com/* wrote: From: Ryan Trauernicht ryanstudyvo...@gmail.com mailto:ryanstudyvo...@gmail.com Subject: [OSL | CCIE_Voice] IPMA Assistant App To: OSL Group ccie_voice@onlinestudylist.com mailto:ccie_voice@onlinestudylist.com Date: Saturday, February 7, 2009, 6:44 AM When the app boot ups it asks for the IPMA server. In the service params I set the Sub first and Pub as backup. If I put in the IP of the Sub it errors out and said it can not find the IPMA server. Is this normal? Thanks, Ryan Trauernicht __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __
Re: [OSL | CCIE_Voice] IPMA Assistant App
From the way the logs look, I dont think you can put both IP addresses in the IPMA assistant console. If you configure it like CMAC where you put Sub , Pub IPMA assistant application tries to goto that exact host 192.168.187.12 , 192.168.187.11 which obviously is not a valid hostname or IP address. I think you are suppose to only point it at the pub. Not sure those. Thanks for the help so far! On Sat, Feb 7, 2009 at 11:40 AM, Ryan Trauernicht ryanstudyvo...@gmail.comwrote: Looks like the port (2912) is not open on the second call manager. I can telnet on that port to the Publisher but not the Subscriber. 67: Thu Feb 05 22:00:47 PST 2009 % ERROR!! Unable to retrieve server locale master file Versions 68: Thu Feb 05 22:00:47 PST 2009 % ServerConnect: Initializing sockets 69: Thu Feb 05 22:00:47 PST 2009 % ServerConnect: Creating a socket connection to host: 192.168.187.12 on port: 2912 70: Thu Feb 05 22:00:48 PST 2009 % ERROR!! ERROR - ServerConnect: caught an exception while initializing the socket java.net.ConnectException: Connection refused: connect 71: Thu Feb 05 22:00:48 PST 2009 % ServerConnect: Initializing sockets 72: Thu Feb 05 22:00:48 PST 2009 % ServerConnect: Creating a socket connection to host: 192.168.187.12 on port: 2912 73: Thu Feb 05 22:00:49 PST 2009 % ERROR!! ERROR - ServerConnect: caught an exception while initializing the socket java.net.ConnectException: Connection refused: connect 74: Thu Feb 05 22:00:49 PST 2009 % ERROR!! Could not connect to any of the servers Attached is the logs. On Sat, Feb 7, 2009 at 11:33 AM, basant yadav basant.ya...@gmail.comwrote: When you open the IPMA Assistant console, click on settings, then go to advanced tab and select Enable trace option. It shows the path there as well where it will save the logs. Reproduce the issue i.e close everything and reopen the IPMA assistant console. It will try to connect with Subscriber server as per settings. If it fails, collect the log file from the specified location. - Basant On Sat, Feb 7, 2009 at 6:29 PM, Ryan Trauernicht ryanstudyvo...@gmail.com wrote: CTI or IPMA logs? IPMA on the Sub do not really have any logs even after changing it to detailed. On Sat, Feb 7, 2009 at 11:25 AM, basant yadav basant.ya...@gmail.comwrote: In that case, Pls collect and send IPMA console logs. Lets see why its failing to connect to SUB. - Basant On Sat, Feb 7, 2009 at 6:20 PM, Ryan Trauernicht ryanstudyvo...@gmail.com wrote: I have tried that as well. I reboot both the sub and pub. Issue still continues. On Sat, Feb 7, 2009 at 11:16 AM, basant yadav basant.ya...@gmail.comwrote: Hi Ryan I just had the same issue a few minutes back. While choosing subscriber as server for IPMA console, getting error cannot find server however when selecting Publisher in place, it worked fine. I fixed it by restarting the Tomcat service on subscriber using services.msc followed by rebooting the subscriber. HTH - Basant On Sat, Feb 7, 2009 at 6:09 PM, Ryan Trauernicht ryanstudyvo...@gmail.com wrote: I have configured it on both servers. Still comes up with the same error. Thanks, Ryan Trauernicht On Sat, Feb 7, 2009 at 6:07 AM, Christian Hennrich christian.hennr...@intact-is.com wrote: hi, have you configured the ipma server on sub and pub. the ipma service parameter need to be configure on both. they are not global. that resolved the problem for me HTH Ryan Trauernicht schrieb: yup. On Fri, Feb 6, 2009 at 8:07 PM, anil batra anil...@yahoo.commailto: anil...@yahoo.com wrote: Is the IPMA service ON Sub --- On *Sat, 2/7/09, Ryan Trauernicht / ryanstudyvo...@gmail.com mailto:ryanstudyvo...@gmail.com/* wrote: From: Ryan Trauernicht ryanstudyvo...@gmail.com mailto:ryanstudyvo...@gmail.com Subject: [OSL | CCIE_Voice] IPMA Assistant App To: OSL Group ccie_voice@onlinestudylist.com mailto:ccie_voice@onlinestudylist.com Date: Saturday, February 7, 2009, 6:44 AM When the app boot ups it asks for the IPMA server. In the service params I set the Sub first and Pub as backup. If I put in the IP of the Sub it errors out and said it can not find the IPMA server. Is this normal? Thanks, Ryan Trauernicht __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __
Re: [OSL | CCIE_Voice] IPMA Assistant App
Since it has kind of been determined that you can only put 1 IP in the IPMA console app which do you generally put in.. Publisher or Subscriber since only one answers on port 2912 at a time. On Sat, Feb 7, 2009 at 11:45 AM, Ryan Trauernicht ryanstudyvo...@gmail.comwrote: From the way the logs look, I dont think you can put both IP addresses in the IPMA assistant console. If you configure it like CMAC where you put Sub , Pub IPMA assistant application tries to goto that exact host 192.168.187.12 , 192.168.187.11 which obviously is not a valid hostname or IP address. I think you are suppose to only point it at the pub. Not sure those. Thanks for the help so far! On Sat, Feb 7, 2009 at 11:40 AM, Ryan Trauernicht ryanstudyvo...@gmail.com wrote: Looks like the port (2912) is not open on the second call manager. I can telnet on that port to the Publisher but not the Subscriber. 67: Thu Feb 05 22:00:47 PST 2009 % ERROR!! Unable to retrieve server locale master file Versions 68: Thu Feb 05 22:00:47 PST 2009 % ServerConnect: Initializing sockets 69: Thu Feb 05 22:00:47 PST 2009 % ServerConnect: Creating a socket connection to host: 192.168.187.12 on port: 2912 70: Thu Feb 05 22:00:48 PST 2009 % ERROR!! ERROR - ServerConnect: caught an exception while initializing the socket java.net.ConnectException: Connection refused: connect 71: Thu Feb 05 22:00:48 PST 2009 % ServerConnect: Initializing sockets 72: Thu Feb 05 22:00:48 PST 2009 % ServerConnect: Creating a socket connection to host: 192.168.187.12 on port: 2912 73: Thu Feb 05 22:00:49 PST 2009 % ERROR!! ERROR - ServerConnect: caught an exception while initializing the socket java.net.ConnectException: Connection refused: connect 74: Thu Feb 05 22:00:49 PST 2009 % ERROR!! Could not connect to any of the servers Attached is the logs. On Sat, Feb 7, 2009 at 11:33 AM, basant yadav basant.ya...@gmail.comwrote: When you open the IPMA Assistant console, click on settings, then go to advanced tab and select Enable trace option. It shows the path there as well where it will save the logs. Reproduce the issue i.e close everything and reopen the IPMA assistant console. It will try to connect with Subscriber server as per settings. If it fails, collect the log file from the specified location. - Basant On Sat, Feb 7, 2009 at 6:29 PM, Ryan Trauernicht ryanstudyvo...@gmail.com wrote: CTI or IPMA logs? IPMA on the Sub do not really have any logs even after changing it to detailed. On Sat, Feb 7, 2009 at 11:25 AM, basant yadav basant.ya...@gmail.comwrote: In that case, Pls collect and send IPMA console logs. Lets see why its failing to connect to SUB. - Basant On Sat, Feb 7, 2009 at 6:20 PM, Ryan Trauernicht ryanstudyvo...@gmail.com wrote: I have tried that as well. I reboot both the sub and pub. Issue still continues. On Sat, Feb 7, 2009 at 11:16 AM, basant yadav basant.ya...@gmail.com wrote: Hi Ryan I just had the same issue a few minutes back. While choosing subscriber as server for IPMA console, getting error cannot find server however when selecting Publisher in place, it worked fine. I fixed it by restarting the Tomcat service on subscriber using services.msc followed by rebooting the subscriber. HTH - Basant On Sat, Feb 7, 2009 at 6:09 PM, Ryan Trauernicht ryanstudyvo...@gmail.com wrote: I have configured it on both servers. Still comes up with the same error. Thanks, Ryan Trauernicht On Sat, Feb 7, 2009 at 6:07 AM, Christian Hennrich christian.hennr...@intact-is.com wrote: hi, have you configured the ipma server on sub and pub. the ipma service parameter need to be configure on both. they are not global. that resolved the problem for me HTH Ryan Trauernicht schrieb: yup. On Fri, Feb 6, 2009 at 8:07 PM, anil batra anil...@yahoo.commailto: anil...@yahoo.com wrote: Is the IPMA service ON Sub --- On *Sat, 2/7/09, Ryan Trauernicht / ryanstudyvo...@gmail.com mailto:ryanstudyvo...@gmail.com/* wrote: From: Ryan Trauernicht ryanstudyvo...@gmail.com mailto:ryanstudyvo...@gmail.com Subject: [OSL | CCIE_Voice] IPMA Assistant App To: OSL Group ccie_voice@onlinestudylist.com mailto:ccie_voice@onlinestudylist.com Date: Saturday, February 7, 2009, 6:44 AM When the app boot ups it asks for the IPMA server. In the service params I set the Sub first and Pub as backup. If I put in the IP of the Sub it errors out and said it can not find the IPMA server. Is this normal? Thanks, Ryan Trauernicht __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email
[OSL | CCIE_Voice] Sanity Check.. SRST caller name
there is no way to get caller name to out in SRST correct? I know it pulls caller name from the display name field on CM, but I didnt know if phones were suppose to keep that in there config file they reregister to the SRST gateway with. thanks, Ryan Trauernicht
[OSL | CCIE_Voice] IPMA Assistant App
When the app boot ups it asks for the IPMA server. In the service params I set the Sub first and Pub as backup. If I put in the IP of the Sub it errors out and said it can not find the IPMA server. Is this normal? Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] IPMA Assistant App
yup. On Fri, Feb 6, 2009 at 8:07 PM, anil batra anil...@yahoo.com wrote: Is the IPMA service ON Sub --- On *Sat, 2/7/09, Ryan Trauernicht ryanstudyvo...@gmail.com* wrote: From: Ryan Trauernicht ryanstudyvo...@gmail.com Subject: [OSL | CCIE_Voice] IPMA Assistant App To: OSL Group ccie_voice@onlinestudylist.com Date: Saturday, February 7, 2009, 6:44 AM When the app boot ups it asks for the IPMA server. In the service params I set the Sub first and Pub as backup. If I put in the IP of the Sub it errors out and said it can not find the IPMA server. Is this normal? Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] SIP Failover commands
Thank you very much Robert. On Thu, Feb 5, 2009 at 2:54 AM, Robert Schuknecht rschukne...@gmx.dewrote: Ryan, some time ago i found an blog article from Vik, where he explained your scenario. For SIP failover you need the following: sip-ua retry invite 2 timers trying 400 You will find Viks article at this link: http://malhi.net/blog/?p=31 HTH /Robert Ryan Trauernichtryanstudyvo...@gmail.com schrieb am Donnerstag, 5. Februar 2009 um 05:22 in Nachricht 9fa638ba8d82c1ae6ebb60f6e8bb1cf5: For H323 you can have a call up and if CM fails the call stays up. If you have a SIP trunk to an FXS port and that CM fails what commands are needed to allow it to failover. The H323 commands obviously dont work. Thanks, Ryan Trauernicht
[OSL | CCIE_Voice] IPCC and MOH Flash scenario
I got a scenario that I dont think is possible with MOH, but I wanted a sanity check. HQ location and IPCC receives MOH multicast from the CM. That same MOH file is on the flash for SiteB and is used locally so no multicast MOH will go across the WAN. So calls from HQ to SiteB and calls from PSTN you hear MOH just fine (MOH source file 1). IPCC requires a different MOH file to be played. So I drop that MOH in the folder and use it as multicast and apply it to the CTI Ports so callers can hear that new MOH file. Works great for HQ phones and PSTN callers calling into the HQ location. Since multicast is not allowed across the WAN and MOH is on the flash for the IP address of (239.1.1.3, the MOH server is in the HQ DP so G729 IP address to the SiteB). If a caller calls in from SiteB to IPCC and it put into the queue you will never hear MOH since it is the second MOH source file that is trying to play, which is being announced on a different IP address. correct? This scenario is not possible right? Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] VPIM woes - troubleshooting
If you read that document, you can not have CUE and Unity in the same forward zone. If you are going to configure VPIM through all DNS you need 2 forward zones (1 for Unity and 1 for CUE). Example: cisco.com (with hostname Unity.cisco.com) cue.cisco.com (with hostname cue.cue.cisco.com) You need to create a mail exchange for each zone using the parent host. You are trying to use the same forward zone for both CUE and Unity. That will not work. Thanks, Ryan Trauernicht On Tue, Feb 3, 2009 at 5:48 PM, Geochelone geochel...@socal.rr.com wrote: I've got VPIM configured and it works from Unity to CUE. When i go from CUE to Unity it tells me your message could not be delivered to extension 1001 at location 100, the recipient mailbox does not exist If i look at trace networking vpim send in CUE, I don't see any output when I try to send the message. CUE config is as follows: network location id 100 email domain unity.cisco.com name unity end location network location id 300 email domain cue.cisco.com name CUE end location network local location id 300 CUE can correctly resolve unity.cisco.com. Are there any other troubleshooting steps I can look at? i've looked around and some people have had to reinstall Unity Voice Connector, but the online lab racks that I use do not have the installation files. TIA
[OSL | CCIE_Voice] Route Not actually marking packets
Anyone ever see a policy map that is set to mark packets to CS3 but doesnt really actually do it. I am marking on the inbound FA and matching on the outbound serial interface. The policy map for inbound show that is it marking and setting but the outbound interface is matching on EF but not CS3. class-map match-any wan-rtp match ip dscp ef class-map match-any lan-rtp match access-group name mark-rtp match ip dscp ef class-map match-any lan-sccp match ip dscp cs3 match ip dscp af31 match protocol skinny match protocol h323 match protocol sip match protocol mgcp class-map match-any wan-sccp match ip dscp cs3 ! ! policy-map lan-mark class lan-rtp set ip dscp ef class lan-sccp set ip dscp cs3 policy-map wan-edge-hq class wan-rtp priority 124 class wan-sccp bandwidth 19 class class-default fair-queue interface FastEthernet0/0 no ip address speed auto ! interface FastEthernet0/0.103 encapsulation dot1Q 103 ip address 192.168.1.1 255.255.255.0 ip helper-address 192.168.187.11 no snmp trap link-status service-policy input lan-mark interface Serial1/0.101 point-to-point bandwidth 384 ip ospf network point-to-point frame-relay interface-dlci 101 ppp Virtual-Template1 class frts-hq ! interface Virtual-Template1 bandwidth 384 ip address 150.101.102.2 255.255.255.0 ip ospf network point-to-point ppp multilink ppp multilink fragment delay 10 ppp multilink interleave service-policy output wan-edge-hq ip access-list extended mark-rtp permit udp any any range 16384 32767 ! ! map-class frame-relay frts-hq frame-relay cir 364800 frame-relay bc 3648 frame-relay be 0 frame-relay mincir 364800
Re: [OSL | CCIE_Voice] Route Not actually marking packets
Sorry for the Spam. Didnt realize you need to look at the virtual-access and not the serial On Wed, Feb 4, 2009 at 8:46 PM, Ryan Trauernicht ryanstudyvo...@gmail.comwrote: Anyone ever see a policy map that is set to mark packets to CS3 but doesnt really actually do it. I am marking on the inbound FA and matching on the outbound serial interface. The policy map for inbound show that is it marking and setting but the outbound interface is matching on EF but not CS3. class-map match-any wan-rtp match ip dscp ef class-map match-any lan-rtp match access-group name mark-rtp match ip dscp ef class-map match-any lan-sccp match ip dscp cs3 match ip dscp af31 match protocol skinny match protocol h323 match protocol sip match protocol mgcp class-map match-any wan-sccp match ip dscp cs3 ! ! policy-map lan-mark class lan-rtp set ip dscp ef class lan-sccp set ip dscp cs3 policy-map wan-edge-hq class wan-rtp priority 124 class wan-sccp bandwidth 19 class class-default fair-queue interface FastEthernet0/0 no ip address speed auto ! interface FastEthernet0/0.103 encapsulation dot1Q 103 ip address 192.168.1.1 255.255.255.0 ip helper-address 192.168.187.11 no snmp trap link-status service-policy input lan-mark interface Serial1/0.101 point-to-point bandwidth 384 ip ospf network point-to-point frame-relay interface-dlci 101 ppp Virtual-Template1 class frts-hq ! interface Virtual-Template1 bandwidth 384 ip address 150.101.102.2 255.255.255.0 ip ospf network point-to-point ppp multilink ppp multilink fragment delay 10 ppp multilink interleave service-policy output wan-edge-hq ip access-list extended mark-rtp permit udp any any range 16384 32767 ! ! map-class frame-relay frts-hq frame-relay cir 364800 frame-relay bc 3648 frame-relay be 0 frame-relay mincir 364800
[OSL | CCIE_Voice] SIP Failover commands
For H323 you can have a call up and if CM fails the call stays up. If you have a SIP trunk to an FXS port and that CM fails what commands are needed to allow it to failover. The H323 commands obviously dont work. Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
; anil...@yahoo.com I tried same way.It plays greeting only once.I also changed service parameter for Cisco TCD Allow Routing with Unknown Line State to True ,and retried.Call still doesn't end. Kapil, how did you add TP as member in HuntGroup.In my case, it gives error saying that member should be a valid DN on system.I was able to add phone/CTIRP DNs though. On Tue, Jan 27, 2009 at 8:48 PM, Kapil Atrish kapilatr...@hotmail.com wrote: I tried with RP/TP Block this pattern and in that case call stays in queue. AC takes the call out of the queue only when it is routed to a registered end-point that's what I've observed. I'll try to route it to some unallocated number pointing it to the GW and see if it works. Thanks for the input. Date: Tue, 27 Jan 2009 10:39:31 +0100 From: christian.hennr...@intact-is.com To: kapilatr...@hotmail.com CC: gree...@googlemail.com; anil...@yahoo.com; cpar...@cparker.us; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? what about routing to a number CUCM, which does not exist, or even to a PSTN number, which is unallocated? Christian Kapil Atrish schrieb: The requirement is to drop the call within CCM itself. I don't want to use Unity/IPCCX/TCL for this purpose. Date: Tue, 27 Jan 2009 09:16:49 + Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? From: gree...@googlemail.com To: anil...@yahoo.com CC: christian.hennr...@intact-is.com; cpar...@cparker.us; kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com Folks, To get the call to disconnect you can use do the following: Create a CTI RP cfwd all to voicemail. In VM create a CH with the extension number of the CTI RP and configure the greeting to be blank and then after greeting send the caller to hang up. In the ac hunt group config add the CTI RP as the always route member. In acconfig.bat for the annunicator ac pilot set the hold time to be something other than 0 seconds After this time has passed the call will be forwarded to unity and disconnected - you get a little bit of ringing as the call gets to unity which I cant get rid of. 2009/1/27 anil batra anil...@yahoo.com I too tried the way Kapil mentioned and faced same issue as he did. The call from PSTN does it the announcement but the call never gets disonncted, it seems the queue is holdin git for forever. Anyone here has tested this and have some workaround please. --- On *Tue, 1/27/09, Kapil Atrish /kapilatr...@hotmail.com/* wrote: From: Kapil Atrish kapilatr...@hotmail.com Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? To: christian.hennr...@intact-is.com, cpar...@cparker.us Cc: ccie_voice@onlinestudylist.com Date: Tuesday, January 27, 2009, 11:38 AM Chris, Your suspicion is what I've in mind that's why I am trying to avoid using Unity/IPCCX/TCL. I've tested AC workaround and its working for me but couple of catches. First of all, the file is in form of MOH and not annunciator which was the original requirement of the question. Secondly, I am not able to disconnect the call. The message keeps on playing until caller drops the call. thanks, Kapil Atrish Date: Mon, 26 Jan 2009 18:57:28 +0100 From: christian.hennr...@intact-is.com To: cpar...@cparker.us CC: ryanstudyvo...@gmail.com; kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? Hi, what about having a MoH File, that is playing the message to the caller. MoH file is played in the AC Hunt group with queueing activated and no AC operators logged in. So you would use only CUCM to play the message. I have not tested that idea, but it might be workable. As far as there is nothing stated, which prevents you from using Unity, I would use Unity. Regards Chris Parker schrieb: The only thing that makes me suspicious about using Unity to play the announcement is that this requirement was listed under the Media section. This leads me to believe they want you to use the annunciator. Otherwise wouldn't it be under the Voicemail/Unity section? Regardless I don't think you can do it any other way unless you hairpin the call through Unity to send the call to the annunciator since the VM ports are skinny registrations. Chris Ryan Trauernicht wrote: That is what I thought but I opened a TAC case and they claim you can, but cant figure out how. Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] mac address of mtp
show int faX/X On Tue, Feb 3, 2009 at 7:58 AM, omar itani ram...@live.com wrote: hi guys 1-how to find the mac address of fastethernet mtp..? -- See all the ways you can stay connected to friends and familyhttp://www.microsoft.com/windows/windowslive/default.aspx
Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
I've observed. I'll try to route it to some unallocated number pointing it to the GW and see if it works. Thanks for the input. Date: Tue, 27 Jan 2009 10:39:31 +0100 From: christian.hennr...@intact-is.com To: kapilatr...@hotmail.com CC: gree...@googlemail.com; anil...@yahoo.com; cpar...@cparker.us; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? what about routing to a number CUCM, which does not exist, or even to a PSTN number, which is unallocated? Christian Kapil Atrish schrieb: The requirement is to drop the call within CCM itself. I don't want to use Unity/IPCCX/TCL for this purpose. Date: Tue, 27 Jan 2009 09:16:49 + Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? From: gree...@googlemail.com To: anil...@yahoo.com CC: christian.hennr...@intact-is.com; cpar...@cparker.us; kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com Folks, To get the call to disconnect you can use do the following: Create a CTI RP cfwd all to voicemail. In VM create a CH with the extension number of the CTI RP and configure the greeting to be blank and then after greeting send the caller to hang up. In the ac hunt group config add the CTI RP as the always route member. In acconfig.bat for the annunicator ac pilot set the hold time to be something other than 0 seconds After this time has passed the call will be forwarded to unity and disconnected - you get a little bit of ringing as the call gets to unity which I cant get rid of. 2009/1/27 anil batra anil...@yahoo.com I too tried the way Kapil mentioned and faced same issue as he did. The call from PSTN does it the announcement but the call never gets disonncted, it seems the queue is holdin git for forever. Anyone here has tested this and have some workaround please. --- On *Tue, 1/27/09, Kapil Atrish /kapilatr...@hotmail.com/* wrote: From: Kapil Atrish kapilatr...@hotmail.com Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? To: christian.hennr...@intact-is.com, cpar...@cparker.us Cc: ccie_voice@onlinestudylist.com Date: Tuesday, January 27, 2009, 11:38 AM Chris, Your suspicion is what I've in mind that's why I am trying to avoid using Unity/IPCCX/TCL. I've tested AC workaround and its working for me but couple of catches. First of all, the file is in form of MOH and not annunciator which was the original requirement of the question. Secondly, I am not able to disconnect the call. The message keeps on playing until caller drops the call. thanks, Kapil Atrish Date: Mon, 26 Jan 2009 18:57:28 +0100 From: christian.hennr...@intact-is.com To: cpar...@cparker.us CC: ryanstudyvo...@gmail.com; kapilatr...@hotmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable? Hi, what about having a MoH File, that is playing the message to the caller. MoH file is played in the AC Hunt group with queueing activated and no AC operators logged in. So you would use only CUCM to play the message. I have not tested that idea, but it might be workable. As far as there is nothing stated, which prevents you from using Unity, I would use Unity. Regards Chris Parker schrieb: The only thing that makes me suspicious about using Unity to play the announcement is that this requirement was listed under the Media section. This leads me to believe they want you to use the annunciator. Otherwise wouldn't it be under the Voicemail/Unity section? Regardless I don't think you can do it any other way unless you hairpin the call through Unity to send the call to the annunciator since the VM ports are skinny registrations. Chris Ryan Trauernicht wrote: That is what I thought but I opened a TAC case and they claim you can, but cant figure out how. Thanks, Ryan Trauernicht On Mon, Jan 26, 2009 at 3:21 AM, Juan juan.c...@gmail.com mailto:juan.c...@gmail.com wrote: I remember reading in the SRND that you can only engage the annunciator for SCCP devices if I remember correctly - so not to the PSTN. cheers, Juan On Sun, Jan 25, 2009 at 5:48 PM, Ryan Trauernicht ryanstudyvo...@gmail.com mailto:ryanstudyvo...@gmail.com wrote: Not sure why you are going through all that trouble and not just sending it to unity as a call handler and hang up after message played. I don't know how to play an ANN from
[OSL | CCIE_Voice] Extracting Tar locally on flash
anyone ever extract a tar file that was copied over to the flash and then I want to extract is locally. I get the following error: SiteB_RTR#archive tar /xtract flash:cme-gui-3.4.0.0.tar flash: extracting admin_user.html (4308 bytes) %Error opening flash:/admin_user.html (Device in exclusive use)
[OSL | CCIE_Voice] CME DSP error
I am using PVDM2's, though I know they are PVDM1's on the lab. I create my PRI and bring up 4 channels. Configure CME completely with CUE and then add in a transcoder. When I hit the max sessions ? I have an option to do up to 6. So I want to max out my router and set it to 6. When I make an inbound call I get fast busy and outbound I get 1/2 ring and then busy. The router throws the following error: *Feb 3 02:35:42.419: %FLEXDSPRM-5-OUT_OF_RESOURCES: No dsps found either locally or globally. I am just wondering if what you can do to get around with. If I drop my max sessions down to 4 I am all set, but I want to max out my router. Thanks, Ryan Trauernicht
[OSL | CCIE_Voice] Trying MLPoFR again
I didnt get a response on any of this... does anyone have an opinion on this? I know this have been talked about many many times... even I have replied and posted about it, but has anyone heard a stance from IPExpert whether MLPoFR with LFI is 13 (like the SRND kind of states) or does that 13 value not include FR layer 2 so we would move it up another 4 to 17bytes for layer 2? Thanks, Ryan Trauernicht Sorry for the SPAM!
Re: [OSL | CCIE_Voice] B-ACD VoiceMail
That is incorrect. The param voice-mail command is the voicemail box that BACD goes to after the param max-time-call-retry XX value has been reached if no one is available. You do not need to put the CUE pilot number anywhere. For example, CUE Pilot could be 5999 and you want BACD to goto 5888 GDM you just put in param voice-mail 5888. Not sure how it knows what CUE pilot is, but it just works. Thanks, Ryan Trauernicht On Fri, Jan 30, 2009 at 6:48 AM, Bradley King collinsda...@gmail.comwrote: I could be wrong, but from what I got from reading the documentation on BACD is that they param voicemail 5000 sends the caller, that is queued, to voice mail after the retry timer for the que to connect the called to the hunt group expires. You have to create a voice mail box in CUE for the pilot number of you BACD, and I guess it depends on what is asked of you to do, but you could make a General Mail Box with the members of the hunt group, and assign the BACD pilot number to this, which will light all MWI on the hunt group members. Brad On Thu, Jan 29, 2009 at 12:00 PM, ccie_voice-requ...@onlinestudylist.comwrote: Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. B-ACD VoiceMail (kamal yousaf) 2. Re: B-ACD VoiceMail (Kumar, Narinder) 3. Re: B-ACD VoiceMail (marwa) 4. Re: B-ACD VoiceMail (kamal yousaf) -- Message: 1 Date: Thu, 29 Jan 2009 22:07:26 +1100 From: kamal yousaf lovingprin...@gmail.com Subject: [OSL | CCIE_Voice] B-ACD VoiceMail To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Message-ID: 34cbd5ac0901290307k5dca3a98yea46bb0812880...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 If B-ACD script causes call to be sent to VoiceMail number defined using 'param voicemail 5000' , which mailbox is the call routed to ? Is it Pilot Point OR HuntGroup Number ? -- next part -- An HTML attachment was scrubbed... URL: http://onlinestudylist.com/pipermail/ccie_voice/attachments/20090129/735b648e/attachment.html -- Message: 2 Date: Thu, 29 Jan 2009 23:30:48 +1100 From: Kumar, Narinder narinder.ku...@uxcg.com.au Subject: Re: [OSL | CCIE_Voice] B-ACD VoiceMail To: kamal yousaf lovingprin...@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Message-ID: 9e7dd48644dd594da5ff12ffa0d2dbe231a6f44...@exmsyd01.aus.local Content-Type: text/plain; charset=us-ascii I don't think 'param voicemail 5000' plays major roll ( I need to study more about BACD) Anyway if you want the BACD call to route to voice mail than you define in the BACD under the number of hunt groups. And define one of the hunt group number as ur voice main pilot ( You won't be achieving much by doing that except you will hear the greeting) You can define a ephone-dn on the CME box forward that ephone-dn number to CUE and create a mailbox for that ephone-dn number. Cheers Narinder From: ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] On Behalf Of kamal yousaf Sent: Thursday, 29 January 2009 10:07 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] B-ACD VoiceMail If B-ACD script causes call to be sent to VoiceMail number defined using 'param voicemail 5000' , which mailbox is the call routed to ? Is it Pilot Point OR HuntGroup Number ? CONFIDENTIALITY - The information contained in this electronic mail message is confidential and is intended solely for the addressee(s). If you are not an authorised recipient of this message please contact Getronics Australia immediately by reply email and destroy/delete this message from your computer. Any unauthorised form of reproduction of this message, or part thereof, is strictly prohibited. DISCLAIMER - Unless specifically indicated otherwise, the views and opinions expressed in this email are those of the sender and not Getronics Australia. While we endeavour to protect our network from computer viruses, Getronics Australia does not warrant that this email or any attachments are free of viruses or any other defects or errors. It is the duty of the recipient to virus scan and otherwise test any information contained in this email before loading onto any computer system. -- next part -- An HTML attachment was scrubbed
Re: [OSL | CCIE_Voice] NTP
Thank you for the reply Mark. What exactly is the Fudge line for My drift file also only has a number in it (23.121). I have never really understood what the number is for. My ntp.config file looks just like below: server 192.168.187.11 # Set Local Clock to Authoritive Time Source -- NTP source IP driftfile C:\WINNT\system32\drivers\etc\ntp.drift # path for drift file I took out the fudge command and I cant remember what was in it untill i rebuild my VM box again tonight, but what should the fudge command be? thanks, Ryan Trauernicht I have set my BR1 to my HQ router (which is the master) and it sync's just fine. On Fri, Jan 30, 2009 at 7:39 AM, Mark Snow ms...@ipexpert.com wrote: Ryan, Try everything again except in your step 5, don't delete the fudge line of code. Then restart the the NTP service however without running the NTPdate.exe. ntpdate.exe and the ntp.conf/ntp service are mutually exclusive form one another. Also, instead of setting your system clock to within 10 mins of the correct time on your ntp master router- set it to an entirely different hour and maybe even year. Every time you stop and start the ntp service, ntp will attempt to update. BTW, let's say your NTP Master router is the HQ router, did you ever try setting up say a BR1 router to be a NTP client to see if it syncs properly with the master first - to make sure the problem doesn't lie with the master instead of the UCM server? If not - be sure to try that first. Cheers, Mark SnowSr Technical Instructor IPexpert, Inc. Sent from my iPhone On Jan 30, 2009, at 8:22 AM, Chris Parker cpar...@cparker.us wrote: Did you set your driftfile in ntp.conf? -- From: Ryan Trauernicht ryanstudyvo...@gmail.com Sent: Thursday, January 29, 2009 11:39 PM To: OSL Group ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] NTP I know this topic has been covered as well... so sorry again for the spam. I have never gotten the NTP on CM to work properly. The process I have always followed is below: 1. StartRunServices.msc 2. Verify the Windows Network Time is disabled 3. Stop Network Time Protocol (leave as automatic) 4. Set local CM time to close to real time (within 10 mins) 5. Update ntp.config (c:\winnt\system32\drivers\etc) with the server x.x.x.x of the NTP server and delete the fudge item in the text file)( 5. open up command prompt and navigate to c:\program files\cisco\xntp 6. run ntpdate.exe x.x.x.x 7. Start Network Time Protocol Once this process is done, CM updates just fine untill it is rebooted. Once it is rebooted my CM goes back to a time I don't know where it is pulling it from. It is a dedicated box to CM so not a VMware. I have lost points on this and I don't want to lose them again any ideas what is missing from this process? thanks! Ryan
[OSL | CCIE_Voice] MLPoFR
I know this have been talked about many many times... even I have replied and posted about it, but has anyone heard a stance from IPExpert whether MLPoFR with LFI is 13 (like the SRND kind of states) or does that 13 value not include FR layer 2 so we would move it up another 4 to 17bytes for layer 2? Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] ATA Passthrough
It is common to change the LBRCodec to 1 (for G711ulaw) as well? Thanks, Ryan Trauernicht On Tue, Jan 27, 2009 at 10:48 PM, Ryan Trauernicht ryanstudyvo...@gmail.com wrote: That is what I thought. Thanks Anil. I got Version 4.0 On Tue, Jan 27, 2009 at 10:45 PM, anil batra anil...@yahoo.com wrote: AudioMode = 0x00150015 ConnectMode = 0x9400 is OK --- On *Wed, 1/28/09, Ryan Trauernicht ryanstudyvo...@gmail.com* wrote: From: Ryan Trauernicht ryanstudyvo...@gmail.com Subject: [OSL | CCIE_Voice] ATA Passthrough To: CCIE Voice Maillist ccie_voice@onlinestudylist.com Date: Wednesday, January 28, 2009, 9:42 AM I have an older version of IPExpert and it said for setting Passthrough on an ATA186 set the following parameters: AudioMode = 0x00140014 ConnectMode = 0x0400 though the following document tells me something different AudioMode = 0x00150015 ConnectMode = 0x9400 http://www.cisco.com/en/US/docs/voice_ip_comm/cata/186_188/2_15_ms/english/administration/guide/sccp/sccpaapd.html Anyone confirm which one should be right? Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] Need some help with IPCC
Have you assigned a skill to the IPCC agent? If so you can you upload you script? Thanks, Ryan Trauernicht On Wed, Jan 28, 2009 at 10:56 PM, Scott ODonnell scott.odonn...@gmail.comwrote: I'm working on an IPCC script that works up to the point where the CONNECT step is executed.I can see while debugging the script that a user variable is populated correctly. When the connect step runs, the agent goes from a reserved state to not ready and the failed branch is followed in the script. Can someone point towards how to troubleshoot? Scott
[OSL | CCIE_Voice] ATA Passthrough
I have an older version of IPExpert and it said for setting Passthrough on an ATA186 set the following parameters: AudioMode = 0x00140014 ConnectMode = 0x0400 though the following document tells me something different AudioMode = 0x00150015 ConnectMode = 0x9400 http://www.cisco.com/en/US/docs/voice_ip_comm/cata/186_188/2_15_ms/english/administration/guide/sccp/sccpaapd.html Anyone confirm which one should be right? Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] ATA Passthrough
That is what I thought. Thanks Anil. I got Version 4.0 On Tue, Jan 27, 2009 at 10:45 PM, anil batra anil...@yahoo.com wrote: AudioMode = 0x00150015 ConnectMode = 0x9400 is OK --- On *Wed, 1/28/09, Ryan Trauernicht ryanstudyvo...@gmail.com* wrote: From: Ryan Trauernicht ryanstudyvo...@gmail.com Subject: [OSL | CCIE_Voice] ATA Passthrough To: CCIE Voice Maillist ccie_voice@onlinestudylist.com Date: Wednesday, January 28, 2009, 9:42 AM I have an older version of IPExpert and it said for setting Passthrough on an ATA186 set the following parameters: AudioMode = 0x00140014 ConnectMode = 0x0400 though the following document tells me something different AudioMode = 0x00150015 ConnectMode = 0x9400 http://www.cisco.com/en/US/docs/voice_ip_comm/cata/186_188/2_15_ms/english/administration/guide/sccp/sccpaapd.html Anyone confirm which one should be right? Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
That is what I thought but I opened a TAC case and they claim you can, but cant figure out how. Thanks, Ryan Trauernicht On Mon, Jan 26, 2009 at 3:21 AM, Juan juan.c...@gmail.com wrote: I remember reading in the SRND that you can only engage the annunciator for SCCP devices if I remember correctly - so not to the PSTN. cheers, Juan On Sun, Jan 25, 2009 at 5:48 PM, Ryan Trauernicht ryanstudyvo...@gmail.com wrote: Not sure why you are going through all that trouble and not just sending it to unity as a call handler and hang up after message played. I don't know how to play an ANN from a PSTN call, I have engaged TAC and they are still working on it and they can't even figure it out right now. Any ideas? Thanks, Ryan Trauernicht On Sun, Jan 25, 2009 at 3:42 AM, Kapil Atrish kapilatr...@hotmail.comwrote: Hi list, Following I did: Create a new MOH Audio Source using AAExtnOutOfService.wav. Prompt available inside Wfavvid folder Create a TP covering all unassigned DNs for example: 11xx, do Called party Xform to 1155 Create a AC Pilot 1155, give any DP say: ANN_PSTN AC Hunt-GroupGive any AC user. No need to login to Attendant Console. Run acconfig.batEnable Queuing Inside DP: ANN_PSTN give User Hold MOH Source as AAExtnOutOfService.wav. Now, whenever you dial any unassigned number withing range 11xx, you'll hear AAExtnOutOfService.wav but the problem is that I am not able to make the PSTN call drop. I tried routing calls to TP inside AC Hunt-GroupAlways Route member is TPTP has Block Pattern --Not working. AAExtnOutOfService.wavkeeps on playing. I tried routing calls to Route-Point (Always Route Member) inside AC Hunt-GroupCTI_RP has Forward all to TPTP has Block Pattern --Not working. AAExtnOutOfService.wavkeeps on playing. I tried routing calls to a registered Phone DN as Always Route MemberForward all to TPTP has Block Pattern --Not working. AAExtnOutOfService.wavkeeps on playing. Can someone help me achieve call drop here without using IPCCX/Unity/TCL? Thanks, Kapil Atrish -- Rediscover the magic of Windows WIN a Windows Vista laptop Windows mobile phone at www.windowsandme.com Try it now!http://www.windowsandme.com
Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
nevermind scratch that Next sentence in that document... It is capable of sending multiple one-way RTP streams to devices such as Cisco IP phones or gateways, and it uses SCCP messages to establish the RTP stream. The device must be capable of SCCP to utilize this feature. since gateways are MGCP or H323 they will not play ANN. But since VG224, VG248, etc... under the gateway page can be SCCP that is why they state gateways in the SRND. Is that correct mark? Thanks, Ryan Trauernicht On Mon, Jan 26, 2009 at 10:47 AM, Ryan Trauernicht ryanstudyvo...@gmail.com wrote: That is what I thought... but page 197 of CM SRND: It is capable of sending multiple one-way RTP streams to devices such as Cisco IP phones or gateways, and it uses SCCP messages to establish the RTP stream. On Mon, Jan 26, 2009 at 10:43 AM, Mark Snow ms...@ipexpert.com wrote: Juan is correct. You cannot play ANN to a GW of any sort, only to SCCP devices. Mark SnowSr Technical Instructor IPexpert, Inc. Sent from my iPhone On Jan 26, 2009, at 11:23 AM, Ryan Trauernicht ryanstudyvo...@gmail.com wrote: That is what I thought but I opened a TAC case and they claim you can, but cant figure out how. Thanks, Ryan Trauernicht On Mon, Jan 26, 2009 at 3:21 AM, Juan juan.c...@gmail.com juan.c...@gmail.com wrote: I remember reading in the SRND that you can only engage the annunciator for SCCP devices if I remember correctly - so not to the PSTN. cheers, Juan On Sun, Jan 25, 2009 at 5:48 PM, Ryan Trauernicht ryanstudyvo...@gmail.com ryanstudyvo...@gmail.com wrote: Not sure why you are going through all that trouble and not just sending it to unity as a call handler and hang up after message played. I don't know how to play an ANN from a PSTN call, I have engaged TAC and they are still working on it and they can't even figure it out right now. Any ideas? Thanks, Ryan Trauernicht On Sun, Jan 25, 2009 at 3:42 AM, Kapil Atrish kapilatr...@hotmail.com kapilatr...@hotmail.com wrote: Hi list, Following I did: Create a new MOH Audio Source using AAExtnOutOfService.wav. Prompt available inside Wfavvid folder Create a TP covering all unassigned DNs for example: 11xx, do Called party Xform to 1155 Create a AC Pilot 1155, give any DP say: ANN_PSTN AC Hunt-GroupGive any AC user. No need to login to Attendant Console. Run acconfig.batEnable Queuing Inside DP: ANN_PSTN give User Hold MOH Source as AAExtnOutOfService.wav. Now, whenever you dial any unassigned number withing range 11xx, you'll hear AAExtnOutOfService.wav but the problem is that I am not able to make the PSTN call drop. I tried routing calls to TP inside AC Hunt-GroupAlways Route member is TPTP has Block Pattern --Not working. AAExtnOutOfService.wavkeeps on playing. I tried routing calls to Route-Point (Always Route Member) inside AC Hunt-GroupCTI_RP has Forward all to TPTP has Block Pattern --Not working. AAExtnOutOfService.wavkeeps on playing. I tried routing calls to a registered Phone DN as Always Route MemberForward all to TPTP has Block Pattern --Not working. AAExtnOutOfService.wavkeeps on playing. Can someone help me achieve call drop here without using IPCCX/Unity/TCL? Thanks, Kapil Atrish -- Rediscover the magic of Windows WIN a Windows Vista laptop Windows mobile phone at http://www.windowsandme.comwww.windowsandme.com Try it now! http://www.windowsandme.com
Re: [OSL | CCIE_Voice] Antw: Re: Fw: Re: VPIM between CUE and Unity using IPaddresses
I have never configured a smart host and I only setup VPIM with DNS on the Unity side. I do not have a forward zone for CUE. Primary zone has name of Unity and domain of ccievoice.com Delivery zone is cue (hostname) and domain of IP address of CUE I also create a mail Exchanger in the forward zone of Unity. On CUE add in the DNS server of Unity and domain name is localhost on CUE Unity zone create a network zone for unity and put in location ID of prefix digits. Location name is unity hostname domain name is ccievoice.com on CUE CUE zone location name is cue hostname domain name is ip address of CUE That is all have ever really configured and it works everytime. I have not had a problem with VPIM. Thanks, Ryan Trauernicht 2009/1/26 o Ninja scarlo...@hotmail.com Robert, Did you remember to create a host inside of Unity´s DNS with the CUE´s ip address ? I followed Steve´s instructions with your´s ip address idea. -- Diversão em dobro: compartilhe fotos enquanto conversa usando o Windows Live Messenger.http://www.microsoft.com/windows/windowslive/messenger.aspx
Re: [OSL | CCIE_Voice] Annunciator to PSTN - Will it be acceptable?
Not sure why you are going through all that trouble and not just sending it to unity as a call handler and hang up after message played. I don't know how to play an ANN from a PSTN call, I have engaged TAC and they are still working on it and they can't even figure it out right now. Any ideas? Thanks, Ryan Trauernicht On Sun, Jan 25, 2009 at 3:42 AM, Kapil Atrish kapilatr...@hotmail.comwrote: Hi list, Following I did: Create a new MOH Audio Source using AAExtnOutOfService.wav. Prompt available inside Wfavvid folder Create a TP covering all unassigned DNs for example: 11xx, do Called party Xform to 1155 Create a AC Pilot 1155, give any DP say: ANN_PSTN AC Hunt-GroupGive any AC user. No need to login to Attendant Console. Run acconfig.batEnable Queuing Inside DP: ANN_PSTN give User Hold MOH Source as AAExtnOutOfService.wav. Now, whenever you dial any unassigned number withing range 11xx, you'll hear AAExtnOutOfService.wav but the problem is that I am not able to make the PSTN call drop. I tried routing calls to TP inside AC Hunt-GroupAlways Route member is TPTP has Block Pattern --Not working. AAExtnOutOfService.wavkeeps on playing. I tried routing calls to Route-Point (Always Route Member) inside AC Hunt-GroupCTI_RP has Forward all to TPTP has Block Pattern --Not working. AAExtnOutOfService.wavkeeps on playing. I tried routing calls to a registered Phone DN as Always Route MemberForward all to TPTP has Block Pattern --Not working. AAExtnOutOfService.wavkeeps on playing. Can someone help me achieve call drop here without using IPCCX/Unity/TCL? Thanks, Kapil Atrish -- Rediscover the magic of Windows WIN a Windows Vista laptop Windows mobile phone at www.windowsandme.com Try it now!http://www.windowsandme.com
Re: [OSL | CCIE_Voice] IPMA and EM on the same phone, please help
Anil is correct... this is normal. On Fri, Jan 23, 2009 at 10:53 AM, anil batra anil...@yahoo.com wrote: I beleive it should since you have definesd the softkey template as IPMA Manager on the phone. --- On *Fri, 1/23/09, jeremy co jeremy.coo...@gmail.com* wrote: From: jeremy co jeremy.coo...@gmail.com Subject: [OSL | CCIE_Voice] IPMA and EM on the same phone, please help To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Friday, January 23, 2009, 9:37 PM Hi, I configured IPMA and EM on same phone. (manager is not EM user) when I login with EM user, I still can see Manager template ( rectangle with 4 icons) . Is this normal? I think it should not appear on the phone. jeremy
Re: [OSL | CCIE_Voice] CME Marking scenario and DSCP Marking on CME IP Phones, anyway to change AF31?
That is exactly what I would do. On Fri, Jan 23, 2009 at 6:58 AM, Kumar, Narinder narinder.ku...@uxcg.com.au wrote: What about you mark on the Voice vlan interface on the router same as you did for CUE. *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *jeremy co *Sent:* Friday, 23 January 2009 6:52 AM *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] CME Marking scenario and DSCP Marking on CME IP Phones, anyway to change AF31? Hi, Scenario: Mark all voice signaling traffic to CS3 on BR2. U are not allowed to remark traffic on switch. for CUE , inbound policy should be used , correct me if I'm wrong. Ras, under dial peer can be specified as CS3 How about ephones? default is AF31, any way to change it? The only way I know to accomplish this scenario is to mark at the wan interface on router, Anything else remain to mark on BR2? Jeremy -- CONFIDENTIALITY - The information contained in this electronic mail message is confidential and is intended solely for the addressee(s). If you are not an authorised recipient of this message please contact Getronics Australia immediately by reply email and destroy/delete this message from your computer. Any unauthorised form of reproduction of this message, or part thereof, is strictly prohibited. DISCLAIMER - Unless specifically indicated otherwise, the views and opinions expressed in this email are those of the sender and not Getronics Australia. While we endeavour to protect our network from computer viruses, Getronics Australia does not warrant that this email or any attachments are free of viruses or any other defects or errors. It is the duty of the recipient to virus scan and otherwise test any information contained in this email before loading onto any computer system.
[OSL | CCIE_Voice] Calling Name CMEGKCM
Any reason why a CME phone that calls to the GK to a CM phone I do not see caller name. CME phone sees name and number of CM phone CM phone only sees number of the CME phone... no caller name. Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] Calling Name CMEGKCM
Dial-peer on the CME side dial-peer voice 5000 voip translation-profile incoming CMInbound destination-pattern [23]...$ session target ras incoming called-number 852T tech-prefix 1 dtmf-relay h245-alphanumeric ip qos dscp cs3 signaling no vad CME Phone making the call ephone-dn 2 dual-line number 4002 no-reg primary description 24024002 name SiteC Phone2 call-forward max-length 0 call-forward busy 4111 call-forward noan 4111 timeout 8 On Wed, Jan 21, 2009 at 8:58 PM, anil batra anil...@yahoo.com wrote: It should I just checked...I am sure you have it deinfed it on ephone-dn, it takes it from there --- On *Thu, 1/22/09, Ryan Trauernicht ryanstudyvo...@gmail.com* wrote: From: Ryan Trauernicht ryanstudyvo...@gmail.com Subject: [OSL | CCIE_Voice] Calling Name CMEGKCM To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Thursday, January 22, 2009, 8:08 AM Any reason why a CME phone that calls to the GK to a CM phone I do not see caller name. CME phone sees name and number of CM phone CM phone only sees number of the CME phone... no caller name. Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] Calling Name CMEGKCM
It is consistant on the CM side. If i take a CM phone and dial a CME phone I only see the number I am calling as well. On Wed, Jan 21, 2009 at 9:00 PM, Ryan Trauernicht ryanstudyvo...@gmail.comwrote: Dial-peer on the CME side dial-peer voice 5000 voip translation-profile incoming CMInbound destination-pattern [23]...$ session target ras incoming called-number 852T tech-prefix 1 dtmf-relay h245-alphanumeric ip qos dscp cs3 signaling no vad CME Phone making the call ephone-dn 2 dual-line number 4002 no-reg primary description 24024002 name SiteC Phone2 call-forward max-length 0 call-forward busy 4111 call-forward noan 4111 timeout 8 On Wed, Jan 21, 2009 at 8:58 PM, anil batra anil...@yahoo.com wrote: It should I just checked...I am sure you have it deinfed it on ephone-dn, it takes it from there --- On *Thu, 1/22/09, Ryan Trauernicht ryanstudyvo...@gmail.com* wrote: From: Ryan Trauernicht ryanstudyvo...@gmail.com Subject: [OSL | CCIE_Voice] Calling Name CMEGKCM To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Thursday, January 22, 2009, 8:08 AM Any reason why a CME phone that calls to the GK to a CM phone I do not see caller name. CME phone sees name and number of CM phone CM phone only sees number of the CME phone... no caller name. Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] Preserving calls?
You can not preserve calls from MGCP to SRST. In the IOS of the code on the lab I dont think you can preserve the calls from H323 to SRST (even though they are 1 in the same) Thanks, Ryan Trauernicht On Tue, Jan 20, 2009 at 1:52 PM, kamal yousaf lovingprin...@gmail.comwrote: Is it possible to preserve calls from Br1 MGCP gateway when it falls back into SRST mode ? I know calls can be preserved while using H323 gateway but not using MGCP gateway since L3 binding is terminated from gateway to CCM when falling back to SRST mode and hence calls cannot be preserved ? Any thoughts on this.
Re: [OSL | CCIE_Voice] Preserving calls?
I apologize on the H323. I stand corrected. Kamal you are correct on the H323 preserving calls. MGCP CM Sub goes down but Pub up call preserved. MGCP All CMs go down... call dropped H323 CM Sub goes down but Pub up call preserved. H323 All CMs go down... call preserved Thanks, Ryan Trauernicht On Tue, Jan 20, 2009 at 7:32 PM, James Key j...@jackhenry.com wrote: Ryan is correct on this. MGCP PRI backhaul does NOT support call preservation when switching to SRST and when rehoming to CCM. Once the connection is lost to CCMs, the D-channel will need to terminate on the gateway, so the backhaul is tore down and the D-channel is reset, Call is dropped. With MGCP analog and CAS, calls WILL be maintained. In the case of Switchover, calls will also be preserved. James Key -- *From:* ccie_voice-boun...@onlinestudylist.com [ ccie_voice-boun...@onlinestudylist.com] On Behalf Of kamal yousaf [ lovingprin...@gmail.com] *Sent:* Tuesday, January 20, 2009 5:32 PM *To:* Ryan Trauernicht *Cc:* ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] Preserving calls? Using H323 GW to SRST fallback,yes you can in Lab IOS version: voice service voip h323 no h225 timeout keepalive ! For later IOS releases, voice service voip h323 call preserve ! Now coming back to question , If br1 gateway was an MGCP gw, then on fallback from CCM, calls will be preserved by default. * http://www.cisco.com/en/US/docs/ios/voice/cminterop/configuration/guide/vc_ucm_mgcp_gw.html *http://www.cisco.com/en/US/docs/ios/voice/cminterop/configuration/guide/vc_ucm_mgcp_gw.html Benefits of Cisco Unified Communications Manager Switchover and MGCP Gateway Fallback •Eliminates a potential single point of failure in the VoIP network by allowing you to designate up to two backup Cisco Unified Communications Manager servers. Your MGCP voice gateways can continue working if the primary Cisco Unified Communications Manager server fails. •Ensures greater stability in the voice network by preserving existing connections during a switchover to a backup Cisco Unified Communications Manager server. •Prevents call-processing interruptions or dropped calls in the event of a Cisco Unified Communications Manager or WAN failure. Rgds On Wed, Jan 21, 2009 at 10:11 AM, Ryan Trauernicht ryanstudyvo...@gmail.com wrote: You can not preserve calls from MGCP to SRST. In the IOS of the code on the lab I dont think you can preserve the calls from H323 to SRST (even though they are 1 in the same) Thanks, Ryan Trauernicht On Tue, Jan 20, 2009 at 1:52 PM, kamal yousaf lovingprin...@gmail.comwrote: Is it possible to preserve calls from Br1 MGCP gateway when it falls back into SRST mode ? I know calls can be preserved while using H323 gateway but not using MGCP gateway since L3 binding is terminated from gateway to CCM when falling back to SRST mode and hence calls cannot be preserved ? Any thoughts on this. NOTICE: This electronic mail message and any files transmitted with it are intended exclusively for the individual or entity to which it is addressed. The message, together with any attachment, may contain confidential and/or privileged information. Any unauthorized review, use, printing, saving, copying, disclosure or distribution is strictly prohibited. If you have received this message in error, please immediately advise the sender by reply email and delete all copies.
Re: [OSL | CCIE_Voice] QOS MLP 1 link and FRF Shaping the other verification
Anil you are almost there. If you are doing FRF.12 you need to have the frame-relay fragment command under the map-class. On Mon, Jan 19, 2009 at 4:12 AM, anil batra anil...@yahoo.com wrote: I am little lost on this ( I think it 2AM affect) ... Module QoS, MQC and CBQFQ - all are same thing but different name So my question is when it says - (1) HQ to BR1 we are to use MLP with LFI (2) HQ to BR2 we are to use FRF.12 Then I will configure MLP with LFI for (1) with frame-relay traffic-shaping command on physical interface And for (2) what will be my configuration, shall it something like this - class-map match all media match ip dscp ef class-map match sig match ip dscp cs3 ! ! policy-map llq class media priority 60 class sig bandwidth 8 class class-default fair-queue ! ! map-class frame-relay frts frame-relay cir 729000 frame-relay mincir 729000 frame-relay bc 7290 frame-relay be 729000 ! ! interface serial 0/0/0:0 frame-relay traffic-shaping ! !interface serial 0/0/00.1 bandwidth 768 frame-relay dlci 101 ip address 162.45.10.101 class frts ! ! thx for your help. //anil --- On *Wed, 1/14/09, Vik Malhi vma...@ipexpert.com* wrote: From: Vik Malhi vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] QOS MLP 1 link and FRF Shaping the other verification To: Ryan Trauernicht ryanstudyvo...@gmail.com, anil...@yahoo.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Wednesday, January 14, 2009, 12:54 PM Agree- possible if you have separate physical interfaces at the hub site but if its the same physical interface then ask the Proctor what he has been smoking. -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: *vma...@ipexpert.com * Join our free online support and peer group communities: *http://www.IPexpert.com/communities *IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- *From: *Ryan Trauernicht ryanstudyvo...@gmail.com *Date: *Mon, 12 Jan 2009 18:22:42 -0600 *To: *anil...@yahoo.com *Cc: *Vik Malhi vma...@ipexpert.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com *Subject: *Re: [OSL | CCIE_Voice] QOS MLP 1 link and FRF Shaping the other verification You will not be asked that. That can not be done. Option 2 needs to be configured as traditional method. thanks, Ryan Trauernicht On Mon, Jan 12, 2009 at 6:17 PM, anil batra anil...@yahoo.com wrote: What if we are to configure in the scenario I mentioned where it says configure 1. HQ to BR1 we are to use MLP with LFI 2. HQ to BR2 we are to use FRF.12 with MQC-FRTS (CB-Shapping way) In the above scenario, on HQ major( Physica) interface is same. But as you mentioned we should not apply Frame-relay Traffic-shaping command for MQC-FRTS but we will have to apply for MLP. In another words the above scenario should be avoided and we shoudl use Leagacy FRTS only for HQ to BR2. That means MLP and MQC sharing same physical interface are mutually exclusive. And hence we shoufl use HQ to BR1 we are to use MLP with LFI and Leagcy FRTS for HQ to BR2 we are to use FRF.12 . -anil --- On *Tue, 1/13/09, Ryan Trauernicht ryanstudyvo...@gmail.com* wrote: From: Ryan Trauernicht ryanstudyvo...@gmail.com Subject: Re: [OSL | CCIE_Voice] QOS MLP 1 link and FRF Shaping the other verification To: anil...@yahoo.com Cc: Vik Malhi vma...@ipexpert.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Tuesday, January 13, 2009, 5:26 AM You need Frame-relay Traffic-shaping command placed on the physical interface (wha tI mean by that is Serial0/0... NOT s0/0.101) So not the PVC when you use FRF.12 traditional way (aka cir, mincir, bc, be) and MLP. You do not put in on the physical interface when you use the CB Shaping way for FRF.12 (aka nested policy-maps) Thanks, Ryan Trauernicht On Mon, Jan 12, 2009 at 5:49 PM, anil batra anil...@yahoo.com wrote: 1. Let's say BR1 to HQ we are to use MLP with LFI but for BR2 to HQ we are to use FRF.12 Fragmentation. What I understand is we will use MLP with LFI between HQ-BR1 with NO frame-realy traffic-shaping command on major interface. Now on BR2 to HQ as we are supposed to use FRF.12 , in this case we will have to use Legacy FRTS for this link but not MQC-FRTS right ??? 2. I am little confused whe do you need to put frame-realy traffic-shaping command on major interface - MLP - I think NO Legacy FRTS - I think NO MQC-FRTS - I think YES regards // anil --- On *Tue, 1/13/09, Vik Malhi vma...@ipexpert.com
Re: [OSL | CCIE_Voice] America numbering plan , does pstn pass 1 or not ?
the 1 is for LD. National has nothing to do with it. Both calls are national. 7 (or 10 digits) are considered local. 1 (LD) - 312 (Area Code) - 555 (CO Number) - 1212 (DID) Local - 3125551212 Local - 5551212 LD - 13125551212 Thanks, Ryan Trauernicht On Mon, Jan 19, 2009 at 6:41 AM, jeremy co jeremy.coo...@gmail.com wrote: Hi, I'm little confused with US numbering plan, since I'm no living in us. Does pstn pass 1 for national calls? Say site a E.164 number is 5031022xxx if some on from PSTN location calls toward our site , PSTN will pass it as 1503xxx- or 503xxx- ? I get confused since I came across scenario that requires in IPCC that match on calling number but number is sth like 1503xxx- . I though 1 would be eliminated by PSTN. Please clarify. Jeremy
Re: [OSL | CCIE_Voice] Multicast from Flash w/ 3725
What about changing it to ip pim sparse-dense-mode I see most people are using dense-mode On Mon, Jan 19, 2009 at 9:19 AM, Kevin Porter kpor...@netelligent.comwrote: If the 3745 has the NM-ESW in it, the no igmp snooping command should help you. I had the exact same thing yesterday…I had a call on hold, dead-air for MoH, entered no igmp snooping and MoH played immediately… -- *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *o Ninja *Sent:* Monday, January 19, 2009 6:59 AM *To:* narinder.ku...@uxcg.com.au; ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] Multicast from Flash w/ 3725 No, I am using just the Pub. I saw in the OSL now that we have to add no igmp snooping and no mgcp timer receive-rtcp. Is that correct ? Could you send me a sh run which has this configuration set ? Thanks for your help. -- From: narinder.ku...@uxcg.com.au To: scarlo...@hotmail.com; ccie_voice@onlinestudylist.com Date: Mon, 19 Jan 2009 23:50:32 +1100 Subject: RE: [OSL | CCIE_Voice] Multicast from Flash w/ 3725 Hi Ninja, Dead air means most probably it is a infrastructure issue. I had the same issue once but my MOH was not in G711 DP/Region and I was using 239.1.1.1 on BR1 instead on 239.1.1.3 Are you using single MOH server of PuB and Sub. For multicast MOH from flash you have to use only one server in the SiteB MRG's . I personally never have tried both Pub and Sub as the multicast ip address will be different and you can't spoof different addresses. Cheers Narinder *From:* o Ninja [mailto:scarlo...@hotmail.com] *Sent:* Monday, 19 January 2009 11:45 PM *To:* Kumar, Narinder; ccie_voice@onlinestudylist.com *Subject:* RE: [OSL | CCIE_Voice] Multicast from Flash w/ 3725 Hi Narinder, Yes, I have all these commands you mentioned. I have got a dead air. Today I have a PL session and I am gonna try this solution again. -- From: narinder.ku...@uxcg.com.au To: scarlo...@hotmail.com; ccie_voice@onlinestudylist.com Date: Mon, 19 Jan 2009 23:36:33 +1100 Subject: RE: [OSL | CCIE_Voice] Multicast from Flash w/ 3725 Do you get tone on hold or dead air... I am assuming you already have Ip source address x.x.x.x port 2000 Max-ephone X Max-dn X Configured under ur call-manager-fallback Do you see any output for sh ccm-manager music when you place the PSTN phone on hold from siteB ? Also multicast moh 239.1.1.1 port 16384 route (Voice VLAN) and (loopback) for ur pots Cheers Narinder *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *o Ninja *Sent:* Monday, 19 January 2009 11:19 PM *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] Multicast from Flash w/ 3725 Hello All, I have done this configuration with no problems at my work and my lab, however I faced some problems using 3725 router, the same we have in the exam. The conf I use I took from IPexpert´s workbook and from their bootcamp: - MoH Server inside of a G711u Region - ip pim disable on 3725 serial interface - *ccm-manager music-on-hold *- callmanager-fallback moh xxx multicast moh 239.1.1.1 ... Are there any further configuration that should be done to avoid any existing bug ? How can I fix this issue and make the conf works ? Thanks in advance. Silvio -- Veja mapas e encontre as melhores rotas para fugir do trânsito com o Live Search Maps! Experimente já!http://www.livemaps.com.br/index.aspx?tr=true -- CONFIDENTIALITY - The information contained in this electronic mail message is confidential and is intended solely for the addressee(s). If you are not an authorised recipient of this message please contact Getronics Australia immediately by reply email and destroy/delete this message from your computer. Any unauthorised form of reproduction of this message, or part thereof, is strictly prohibited. DISCLAIMER - Unless specifically indicated otherwise, the views and opinions expressed in this email are those of the sender and not Getronics Australia. While we endeavour to protect our network from computer viruses, Getronics Australia does not warrant that this email or any attachments are free of viruses or any other defects or errors. It is the duty of the recipient to virus scan and otherwise test any information contained in this email before loading onto any computer system. -- Veja mapas e encontre as melhores rotas para fugir do trânsito com o Live Search Maps! Experimente já!http://www.livemaps.com.br/index.aspx?tr=true -- CONFIDENTIALITY - The information contained in this electronic mail message is confidential and is intended solely for
Re: [OSL | CCIE_Voice] Multicast from Flash w/ 3725
if you are doing it from the flash and you announce both loopback and voice vlan interface in the multicast moh statement you dont need multicast enabled on the remote side. This is ONLY if you are doing MOH from the flash on the remote site. On Mon, Jan 19, 2009 at 9:34 AM, Kevin Porter kpor...@netelligent.comwrote: I have always used ip pim sparse-dense-mode, be sure and apply to Loopback interface and VLAN interface (assuming NM-ESW is installed)… -- *From:* Ryan Trauernicht [mailto:ryanstudyvo...@gmail.com] *Sent:* Monday, January 19, 2009 9:32 AM *To:* Kevin Porter *Cc:* ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] Multicast from Flash w/ 3725 What about changing it to ip pim sparse-dense-mode I see most people are using dense-mode On Mon, Jan 19, 2009 at 9:19 AM, Kevin Porter kpor...@netelligent.com wrote: If the 3745 has the NM-ESW in it, the no igmp snooping command should help you. I had the exact same thing yesterday…I had a call on hold, dead-air for MoH, entered no igmp snooping and MoH played immediately… -- *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *o Ninja *Sent:* Monday, January 19, 2009 6:59 AM *To:* narinder.ku...@uxcg.com.au; ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] Multicast from Flash w/ 3725 No, I am using just the Pub. I saw in the OSL now that we have to add no igmp snooping and no mgcp timer receive-rtcp. Is that correct ? Could you send me a sh run which has this configuration set ? Thanks for your help. -- From: narinder.ku...@uxcg.com.au To: scarlo...@hotmail.com; ccie_voice@onlinestudylist.com Date: Mon, 19 Jan 2009 23:50:32 +1100 Subject: RE: [OSL | CCIE_Voice] Multicast from Flash w/ 3725 Hi Ninja, Dead air means most probably it is a infrastructure issue. I had the same issue once but my MOH was not in G711 DP/Region and I was using 239.1.1.1 on BR1 instead on 239.1.1.3 Are you using single MOH server of PuB and Sub. For multicast MOH from flash you have to use only one server in the SiteB MRG's . I personally never have tried both Pub and Sub as the multicast ip address will be different and you can't spoof different addresses. Cheers Narinder *From:* o Ninja [mailto:scarlo...@hotmail.com] *Sent:* Monday, 19 January 2009 11:45 PM *To:* Kumar, Narinder; ccie_voice@onlinestudylist.com *Subject:* RE: [OSL | CCIE_Voice] Multicast from Flash w/ 3725 Hi Narinder, Yes, I have all these commands you mentioned. I have got a dead air. Today I have a PL session and I am gonna try this solution again. -- From: narinder.ku...@uxcg.com.au To: scarlo...@hotmail.com; ccie_voice@onlinestudylist.com Date: Mon, 19 Jan 2009 23:36:33 +1100 Subject: RE: [OSL | CCIE_Voice] Multicast from Flash w/ 3725 Do you get tone on hold or dead air... I am assuming you already have Ip source address x.x.x.x port 2000 Max-ephone X Max-dn X Configured under ur call-manager-fallback Do you see any output for sh ccm-manager music when you place the PSTN phone on hold from siteB ? Also multicast moh 239.1.1.1 port 16384 route (Voice VLAN) and (loopback) for ur pots Cheers Narinder *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *o Ninja *Sent:* Monday, 19 January 2009 11:19 PM *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] Multicast from Flash w/ 3725 Hello All, I have done this configuration with no problems at my work and my lab, however I faced some problems using 3725 router, the same we have in the exam. The conf I use I took from IPexpert´s workbook and from their bootcamp: - MoH Server inside of a G711u Region - ip pim disable on 3725 serial interface - *ccm-manager music-on-hold *- callmanager-fallback moh xxx multicast moh 239.1.1.1 ... Are there any further configuration that should be done to avoid any existing bug ? How can I fix this issue and make the conf works ? Thanks in advance. Silvio -- Veja mapas e encontre as melhores rotas para fugir do trânsito com o Live Search Maps! Experimente já!http://www.livemaps.com.br/index.aspx?tr=true -- CONFIDENTIALITY - The information contained in this electronic mail message is confidential and is intended solely for the addressee(s). If you are not an authorised recipient of this message please contact Getronics Australia immediately by reply email and destroy/delete this message from your computer. Any unauthorised form of reproduction of this message, or part thereof, is strictly prohibited. DISCLAIMER - Unless specifically indicated otherwise, the views and opinions expressed in this email
[OSL | CCIE_Voice] NBAR
Anyone got thoughts on using NBAR for the lab to mark packets? Best practice in the field is to use access-lists b/c NBAR causes to much processor power, but will you be docked if you just used NBAR for protocols (skinny, h323, mgcp, sip)? Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] Multicast from Flash w/ 3725
ip pim sparse-dense-mode is just a push method for multicast. dense-mode is a pull method. You do not need RP for sparse-dense-mode You do need them for sparse-mode for ip pim sparse-dense-mode you just need to enable multicast and put ip pim sparse-dense-mode on all the routing interfaces you need the multicast to flow across. On Mon, Jan 19, 2009 at 10:06 AM, o Ninja scarlo...@hotmail.com wrote: Hi, Well, I use the ip pim dense-mode because it is the easiest approach if I am not mistaken, with it we dont have to configure rps through the network. From: ccie_voice-requ...@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 35, Issue 166 To: ccie_voice@onlinestudylist.com Date: Mon, 19 Jan 2009 10:46:42 -0500 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: Multicast from Flash w/ 3725 (Kevin Porter) 2. Re: QoS on router causing crash... (Kevin Porter) -- Message: 1 Date: Mon, 19 Jan 2009 09:34:40 -0600 From: Kevin Porter kpor...@netelligent.com Subject: Re: [OSL | CCIE_Voice] Multicast from Flash w/ 3725 To: ccie_voice@onlinestudylist.com Message-ID: cd530e27013ed34e8748c71522e4c22e01c31...@nc-svr-23.netelligent.com Content-Type: text/plain; charset=iso-8859-1 I have always used ip pim sparse-dense-mode, be sure and apply to Loopback interface and VLAN interface (assuming NM-ESW is installed)... From: Ryan Trauernicht [mailto:ryanstudyvo...@gmail.com] Sent: Monday, January 19, 2009 9:32 AM To: Kevin Porter Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Multicast from Flash w/ 3725 What about changing it to ip pim sparse-dense-mode I see most people are using dense-mode On Mon, Jan 19, 2009 at 9:19 AM, Kevin Porter kpor...@netelligent.com wrote: If the 3745 has the NM-ESW in it, the no igmp snooping command should help you. I had the exact same thing yesterday...I had a call on hold, dead-air for MoH, entered no igmp snooping and MoH played immediately... From: ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] On Behalf Of o Ninja Sent: Monday, January 19, 2009 6:59 AM To: narinder.ku...@uxcg.com.au; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Multicast from Flash w/ 3725 No, I am using just the Pub. I saw in the OSL now that we have to add no igmp snooping and no mgcp timer receive-rtcp. Is that correct ? Could you send me a sh run which has this configuration set ? Thanks for your help. From: narinder.ku...@uxcg.com.au To: scarlo...@hotmail.com; ccie_voice@onlinestudylist.com Date: Mon, 19 Jan 2009 23:50:32 +1100 Subject: RE: [OSL | CCIE_Voice] Multicast from Flash w/ 3725 Hi Ninja, Dead air means most probably it is a infrastructure issue. I had the same issue once but my MOH was not in G711 DP/Region and I was using 239.1.1.1 on BR1 instead on 239.1.1.3 Are you using single MOH server of PuB and Sub. For multicast MOH from flash you have to use only one server in the SiteB MRG's . I personally never have tried both Pub and Sub as the multicast ip address will be different and you can't spoof different addresses. Cheers Narinder From: o Ninja [mailto:scarlo...@hotmail.com] Sent: Monday, 19 January 2009 11:45 PM To: Kumar, Narinder; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Multicast from Flash w/ 3725 Hi Narinder, Yes, I have all these commands you mentioned. I have got a dead air. Today I have a PL session and I am gonna try this solution again. From: narinder.ku...@uxcg.com.au To: scarlo...@hotmail.com; ccie_voice@onlinestudylist.com Date: Mon, 19 Jan 2009 23:36:33 +1100 Subject: RE: [OSL | CCIE_Voice] Multicast from Flash w/ 3725 Do you get tone on hold or dead air... I am assuming you already have Ip source address x.x.x.x port 2000 Max-ephone X Max-dn X Configured under ur call-manager-fallback Do you see any output for sh ccm-manager music when you place the PSTN phone on hold from siteB ? Also multicast moh 239.1.1.1 port 16384 route (Voice VLAN) and (loopback) for ur pots Cheers Narinder From: ccie_voice-boun
Re: [OSL | CCIE_Voice] NBAR
Thinking about my class-map to look like the following for marking on the ingress Class-map match-any SCCP Match protocol skinny Match protocol h323 Match protocol mgcp Match protocol sip match ip dscp cs3 match ip dscp af31 Class-map match-any RTP Match protocol rtp audio match ip dscp ef any thoughts? On Mon, Jan 19, 2009 at 10:44 AM, Cyrus cyrus@gmail.com wrote: Ryan I will use it ,it's more easy to use NBAR than access lists. same result. High process utilization does not breaking any lab requirements if nothing specified. I know when it comes to lab exam, picking up the right tool becomes a nightmare ,I'm too fussy about it! :) Cyrus On Tue, Jan 20, 2009 at 3:01 AM, Ryan Trauernicht ryanstudyvo...@gmail.com wrote: Anyone got thoughts on using NBAR for the lab to mark packets? Best practice in the field is to use access-lists b/c NBAR causes to much processor power, but will you be docked if you just used NBAR for protocols (skinny, h323, mgcp, sip)? Thanks, Ryan Trauernicht -- Sirus Moghadasian CCIE #21862 (RS)
Re: [OSL | CCIE_Voice] SRST TCLscript aa
Where did you get those files I do not see them in the following link: http://www.cisco.com/cgi-bin/tablebuild.pl/tclware On Fri, Jan 16, 2009 at 10:13 PM, jeremy co jeremy.coo...@gmail.com wrote: Hi, I came across configuring srst aa task. My understanding is there are two type of tcl used for this matter : its_cisco.2.0.1.0.tcl - require cm-pilot app-b-acd-aa-2.1.0.0.tcl --- there is no cm-pilot configuration for these two are different. Which one will be used in lab exam? Jeremy
[OSL | CCIE_Voice] AA SRST Script
With this version of the lab... is the TCL script and commands the same between CME BACD and SRST AA? Or is there a different script that will be used in the lab version? Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] GK trunk h323 id for pub and sub selection
Cryus is correct. When you start the CCM service it is put in the SQL DB with _1 in the CCM server name. You can verify this if you know your way around SQL DB level. If you look at the CM Group in the SQL you will see the hostname with Pub_1 and Sub_2 assuming you started the CCM service on the Pub first. Just deactivating does not change this. You need a fresh build then. thanks, Ryan Trauernicht On Sat, Jan 17, 2009 at 7:09 PM, Cyrus cyrus@gmail.com wrote: Mark, It depends on which server's CCM service you activated first. If you activate Sub CCM service first ,then your First trunk ID would be _1. And also there is a parameter CTI ID in CCM database that determine actual ID of Trunk ID. By changing to whatever you like ,you can affect Trunk ID numbering. Both CCM services on Pub and Sub should be restarted as well as Trunk to changes take effect. HTH, On Sun, Jan 18, 2009 at 1:49 AM, Mark Snow ms...@ipexpert.com wrote: Jeremy, _1 as a Trunk suffix ALWAYS indicates the first UCM registered to the cluster and therefore ALWAYS will be your Publisher server in amy given cluster. This is non-configurable. Likewise _2 will always be the second UCM in a cluster and thus always a Subscriber server. HTH, Mark SnowSr Technical Instructor IPexpert, Inc. Sent from my iPhone On Jan 17, 2009, at 2:19 AM, jeremy co jeremy.coo...@gmail.com wrote: Agreed, the point is task require me to sub be GK-Trunk_1 and pub GK-Trunk_2 but it ends revers on gatekeeper. reload didn't help. still cannot make them to do it opposite. Jeremy On Sat, Jan 17, 2009 at 5:58 PM, Rogers O. OCHIENG r.ochi...@smoothtel.com r.ochi...@smoothtel.com wrote: They both register with equal priority no matter the who registers first, the suffix does not indicate the order in the CM group. Assuming your extenions in CM are in the range 3XXX Zone prefix 3... gw-priority 10 sub GK-Trunk_1 Zone prefix 3... gw-priority 9 pub GK-Trunk_2 So your sub will always win *From:* ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *anil batra *Sent:* 17 January 2009 06:51 *To:* Kumar, Narinder; jeremy co *Cc:* ccie_voice@onlinestudylist.comccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] GK trunk h323 id for pub and sub selection Stop the CCM on Pub and Sub and then start in same orderworst come worst shut down servers in same order and bring up in same order --- On *Sat, 1/17/09, jeremy co jeremy.coo...@gmail.com jeremy.coo...@gmail.com* wrote: From: jeremy co jeremy.coo...@gmail.comjeremy.coo...@gmail.com Subject: Re: [OSL | CCIE_Voice] GK trunk h323 id for pub and sub selection To: Kumar, Narinder narinder.ku...@uxcg.com.au narinder.ku...@uxcg.com.au Cc: ccie_voice@onlinestudylist.comccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Saturday, January 17, 2009, 9:16 AM Hi, So this happened to me abnormally? how can I change it? Jeremy On Sat, Jan 17, 2009 at 2:39 PM, Kumar, Narinder narinder.ku...@uxcg.com.au narinder.ku...@uxcg.com.au wrote: Jeremy, I don't think you can force the CCM trunks these trunks register with GK dynamically. *From:* ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *jeremy co *Sent:* Saturday, 17 January 2009 12:55 PM *To:* ccie_voice@onlinestudylist.comccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] GK trunk h323 id for pub and sub selection Hi, I configured GK trunk , and it should be registered as : pub GK-Trunk_2 sub GK-Trunk_1 but what I can see under sh gatekeeper endpoints is : pub GK-Trunk_1 sub GK-Trunk_2 So How can I change the oder and force it to use sub as primary (GK-Trunk_1 ) and pub as secondary ( GK-Trunk_2) Jeremy -- CONFIDENTIALITY - The information contained in this electronic mail message is confidential and is intended solely for the addressee(s). If you are not an authorised recipient of this message please contact Getronics Australia immediately by reply email and destroy/delete this message from your computer. Any unauthorised form of reproduction of this message, or part thereof, is strictly prohibited. DISCLAIMER - Unless specifically indicated otherwise, the views and opinions expressed in this email are those of the sender and not Getronics Australia. While we endeavour to protect our network from computer viruses, Getronics Australia does not warrant that this email or any attachments are free of viruses or any other defects or errors. It is the duty of the recipient to virus scan and otherwise test any information contained
Re: [OSL | CCIE_Voice] Marking on Routers !!!
On a side note... I am confused why people are using that type of access-list: acess-list 101 permit tcp any any eq 2000 acess-list 101 permit tcp any any eq 2428 acess-list 101 permit udp any any eq 2427 acess-list 101 permit tcp any eq 1720 any acess-list 101 permit tcp any eq 1718 any acess-list 101 permit udp any eq 1719 any acess-list 101 permit udp any eq 5060 any acess-list 101 permit tcp any eq 5060 any If you are marking on the ingress of the router, why would you be matching on the source port of SIP, RAS and Signaling. You should be doing it in the destination port of those shouldnt you? Thanks, Ryan Trauernicht On Sun, Jan 18, 2009 at 1:07 PM, anil batra anil...@yahoo.com wrote: When it says mark on router ONLY and no configuration be done on switches...do we still need to mark the packets on router using Policy-map. OR specifying ip qos dscp cs3 signal will be sufficinet on BR1 and BR2 voip dial-peers And on HQ we will have to do marking using Policy-map. And if at all we need to mark on BR1 and BR2 what should be the direction. I beleive it shooud be - On BR1 --- class-map match-media match access-group 102 class-map match -sginal match access-group 101 ! ! policy-map mark class media set ip dscp ef class signal set ip dscp cs3 ! ! acess-list 101 permit tcp any any eq 2000 acess-list 101 permit tcp any any eq 2428 acess-list 101 permit udp any any eq 2427 acess-list 101 permit tcp any eq 1720 any acess-list 101 permit tcp any eq 1718 any acess-list 101 permit udp any eq 1719 any acess-list 101 permit udp any eq 5060 any acess-list 101 permit tcp any eq 5060 any ! ! acess-list 102 permit tcp any any range 16384 32767 acess-list 102 permit tcp any range 16384 32767 any ! ! int vlan 240 voice interface vlan service-input mark On BR2 --- class-map match-media match access-group 102 class-map match -sginal match access-group 101 ! ! policy-map mark class media set ip dscp ef class signal set ip dscp cs3 ! ! acess-list 101 permit tcp any any eq 2000 acess-list 101 permit tcp any any eq 2428 acess-list 101 permit udp any any eq 2427 acess-list 101 permit tcp any eq 1720 any acess-list 101 permit tcp any eq 1718 any acess-list 101 permit udp any eq 1719 any acess-list 101 permit udp any eq 5060 any acess-list 101 permit tcp any eq 5060 any ! ! acess-list 102 permit tcp any any range 16384 32767 acess-list 102 permit tcp any range 16384 32767 any ! ! int fa0/0.240 voice subinterface service-input mark -- ON HQ - same as above with port direction in above diraction and.. Vik can you please comment on this one please. regds // anil
Re: [OSL | CCIE_Voice] Gatekeeper: CCM to BR2 call failed: Debug outputand config shown
Tony, you must has the RL configured wrong to prepend the 2#. Even though you say it is there, the call is not prepending it. You will see the following part of the debug: Jan 18 21:05:27.299: //00CABDCA0200/00CABDCA0200/GK/rassrv_get_addrinfo: (3001) Tech-prefix match failed. You should see in the debug 2#3001 and not 3001 I would double check the RL. You can always verify that is your problem by setting up a route list with 2#3001 and just dial that from your phone without adding or striping anything. That will prove it is something on the CM side. Thanks, Ryan Trauernicht On Sun, Jan 18, 2009 at 3:19 PM, Tony reyes treye...@yahoo.com wrote: Thanks Alex, I just checked it and it is there under the RL for using the GK for both the BR1 and HQ? I've also restarted both the CM Pub and Sub and still get the same output from the debug gatek main 10 cmd To me it seems like it is missing something or something is wrong on the BR2 rtr config? But from the show gatek outputs both the CCMs and BR2 rtr show that they are registered with the gatek with the correct Tech prefix of @2# using the show gatek end and show gatek gw cmds? Here is te latest Debug otput: HQ-RTR#debug gatek main 10 HQ-RTR# Jan 18 21:05:27.299: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Jan 18 21:05:27.299: ////GK/gk_rassrv_arq: arqp=0x49058A14,crv=0x2, answerCall=0 Jan 18 21:05:27.299: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC Jan 18 21:05:27.299: //00CABDCA0200/00CABDCA0200/GK/gk_dns_query: No Name servers Jan 18 21:05:27.299: //00CABDCA0200/00CABDCA0200/GK/rassrv_get_addrinfo: (3001) Tech-prefix match failed. Jan 18 21:05:27.299: //00CABDCA0200/00CABDCA0200/GK/rassrv_get_addrinfo: (3001) Matched zone prefix 3 and remainder 001 Jan 18 21:05:27.299: ////GK/gk_rassrv_get_ingress_network: returning default ingress network = 1 Jan 18 21:05:27.299: //00CABDCA0200/00CABDCA0200/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x48DF1464 Jan 18 21:05:27.299: //00CABDCA0200/00CABDCA0200/GK/rassrv_arq_select_viazone: matched zone is CCM-GK, and z_invianamelen=0 Jan 18 21:05:27.299: //00CABDCA0200/00CABDCA0200/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x48DF16D0 Jan 18 21:05:27.299: //00CABDCA0200/00CABDCA0200/GK/rassrv_arq_select_viazone: matched zone is BR2-GK, and z_outvianamelen=0 Jan 18 21:05:27.299: //00CABDCA0200/00CABDCA0200/GK/rassrv_get_addrinfo: No tech prefix Jan 18 21:05:27.299: //00CABDCA0200/00CABDCA0200/GK/rassrv_get_addrinfo: Alias not found Jan 18 21:05:27.299: //00CABDCA0200/00CABDCA0200/GK/rassrv_get_addrinfo: (3001) unknown address and no default technology defined. Jan 18 21:05:27.299: //00CABDCA0200/00CABDCA0200/GK/gk_rassrv_sep_arq: rassrv_get_addrinfo() failed (return code = 0x103) Jan 18 21:05:30.367: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup HQ-RTR# The HQ rtr config gatekeeper zone local CCM-GK cisco.com 172.1.100.1 zone local BR2-GK cisco.com no zone subnet CCM-GK default enable zone subnet CCM-GK 10.1.200.20/32 enable zone subnet CCM-GK 10.1.200.21/32 enable no zone subnet BR2-GK default enable zone subnet BR2-GK 172.1.102.1/32 enable zone prefix CCM-GK 1... gw-priority 10 gk_trunk_2 zone prefix CCM-GK 1... gw-priority 9 gk_trunk_1 zone prefix CCM-GK 2... gw-priority 10 gk_trunk_2 zone prefix CCM-GK 2... gw-priority 9 gk_trunk_1 zone prefix BR2-GK 3... bandwidth interzone zone BR2-GK 32 no shutdown Any ther ideas? Lke i said if I set the default technology prefix, It works? Thanks in advance for the help. Tony -- *From:* Alex alex.arsen...@gmail.com *To:* Tony reyes treye...@yahoo.com; ccie_voice@onlinestudylist.com *Sent:* Sunday, January 18, 2009 1:55:18 PM *Subject:* Re: [OSL | CCIE_Voice] Gatekeeper: CCM to BR2 call failed: Debug outputand config shown Jan 18 18:26:01.870: //006158870500/006158870500/GK/rassrv_get_addrinfo: (3001) Tech-prefix match failed.= You are not prepending 3001 with 2#. Make sure 2# is prepended to 3001 on CCM Route-list/Route-Group level. Rgds Alex - Original Message - *From:* Tony reyes treye...@yahoo.com *To:* ccie_voice@onlinestudylist.com *Sent:* Sunday, January 18, 2009 7:13 PM *Subject:* [OSL | CCIE_Voice] Gatekeeper: CCM to BR2 call failed: Debug outputand config shown Trying to get a calle establsihed from CCM to CME via Gatekeeper. From MGCP BR1 extension 2001 dialed 3001 (73001) at BR2 CME site throuh GK. Please see output and config of HQ and BR2 Rtrs. The call seems to work when I set the default technology prefix, I tought that I didn't need to hae this command? HQ-RTR#debug gatek main 15 calling from extension 2001 to 3001=== HQ-RTR# Jan 18 18:26:01.866: //
Re: [OSL | CCIE_Voice] SRST to VM
I dont think they will not allow you to use that method meaning you should be able to use that work around if you get the RDNIS bug of FF on the redirect. There is another method of vm-integration, but I can never seem to get it to work. Thanks, Ryan Trauernicht On Sun, Jan 18, 2009 at 2:50 PM, Kevin Porter kpor...@netelligent.comwrote: What is the solution for VM integration while in SRST mode if you can't use the DID block at the site where the Voice Mail server is located? Thanks, Kevin
[OSL | CCIE_Voice] IP Agent and IPMA services on 1 phone not working
Anyone ever see it just sitting there requesting for IPMA service. I have checked the port mismatch bug also and changed it to port 8010. Any ideas? Below is the service url. http://192.168.187.11/ma/servlet/MAService?cmd=doPhoneServiceName=#DEVICENAME#
[OSL | CCIE_Voice] Daylights savings
Just wondering if the anyone thinks have been docked points for not putting in the new DST commands into the routers. When you setup NTP on the routers the IOS in the lab doesn't have the most up-to-date daylight savings days since they change in 07 I believe. old command: clock timezone EST -8 clock summer-time EDT recurring ntp server x.x.x.x prefer new commands: clock timezone EST -8 clock summer-time EST recurring 2 Sun Mar 2:00 1 Sun Nov 2:00 ntp server x.x.x.x prefer Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] IP Agent and IPMA services on 1 phone not working
Yeah a reboot just fixed it. IPMA was all up and working, but the service wouldnt go until I bounced CM. Go figured. Thanks Anil On Sun, Jan 18, 2009 at 10:55 PM, anil batra anil...@yahoo.com wrote: what happens when you enter url - http://192.168.187.11/manager/list do you see tomcat servcie window and tried reloading itif you see this pagethat means your pors are good... --- On *Mon, 1/19/09, Ryan Trauernicht ryanstudyvo...@gmail.com* wrote: From: Ryan Trauernicht ryanstudyvo...@gmail.com Subject: [OSL | CCIE_Voice] IP Agent and IPMA services on 1 phone not working To: CCIE Voice Maillist ccie_voice@onlinestudylist.com Date: Monday, January 19, 2009, 6:42 AM Anyone ever see it just sitting there requesting for IPMA service. I have checked the port mismatch bug also and changed it to port 8010. Any ideas? Below is the service url. http://192.168.187.11/ma/servlet/MAService?cmd=doPhoneServiceName=#DEVICENAME#http://192.168.187.11/ma/servlet/MAService?cmd=doPhoneServiceName=#DEVICENAME%23
Re: [OSL | CCIE_Voice] SRST to VM
Karuna, I have never been able to get it to work with the vm-integration no matter what I did. It always went to the opening greeting. Atleast not without changing the dialplan on the PSTN router to except dials that are not NANP (aka 7 digits, 10 digits, and 11 digits). If you ever got it work can you post you config? Thanks, Ryan Trauernicht On Sun, Jan 18, 2009 at 10:55 PM, karuna durai karu...@gmail.com wrote: Hi, Whats your dial plan and are u able to reach HQ site when SRST, you need to configrure , I n you mailbox you need to give alt extn for SRST mode. and below config required at SRST vm-integation pattern trunk busy * FDN pattern trunk no-an * FDN On Mon, Jan 19, 2009 at 2:20 AM, Kevin Porter kpor...@netelligent.comwrote: What is the solution for VM integration while in SRST mode if you can't use the DID block at the site where the Voice Mail server is located? Thanks, Kevin
Re: [OSL | CCIE_Voice] network-clock-select command
It is a must command if you are the user side of the clocking. You must remove it if you are the network side. thanks, Ryan Trauernicht On Fri, Jan 16, 2009 at 8:52 AM, Agh agehac...@gmail.com wrote: Is this command a must? If not, when do we need it? Do we need it even if we source the clock from the line? network-clock-select 1 E1 0/0/0
Re: [OSL | CCIE_Voice] NTP CM
thanks Christian.. I will check tonigth and let you know. On Thu, Jan 15, 2009 at 12:16 PM, Christian Hennrich christian.hennr...@intact-is.com wrote: Hi, I'm using VMware server and I noticed the same behaviour. In VMWare Server you can disable to sync the vm with the host, aka linux vmware server. Since then I haven't had problems any more. This is in one of the sub settings of vmware machine. I do not know, where to find it in the esx box. Ryan Trauernicht schrieb: That did not help Digging alittle bit further into this I see that my CM is actually pulling clock from my ESX box. Not sure why that is happening. Anyone else running ESX for their Call Manager having the same issue? On Wed, Jan 14, 2009 at 10:41 PM, karuna durai karu...@gmail.commailto: karu...@gmail.com wrote: Hi, After editing the ntp.conf file please fo to CMD as C:\Prog file\cisco\xntp ntpdate -b IPADD of NTP pls try this and let me know On Thu, Jan 15, 2009 at 4:33 AM, Ryan Trauernicht ryanstudyvo...@gmail.com mailto:ryanstudyvo...@gmail.com wrote: Any reason why CM will not keep its NTP clock. I have a local router with the following commands: ntp master 3 ntp source loopback0 (IP address is 192.168.187.1) I have edited the c:\WINNT\System32\Drivers\Etc\ntp.config My file looks like: server 192.168.187.1 # Set Local Clock to Authoritive Time Source fudge 192.168.187.1 stratum 5 # Resets Stratum from default 3 to 5 driftfile C:\WINNT\system32\drivers\etc\ntp.drift # path for drift file I stopped the Network Time Protocol service and I run ntpdate.exe 192.168.187.1 command from the cmd. That sets the clock to sync to NTP router just fine. After I reboot CM it goes back to GMT it looks like. I see it trying to sync but it never does. I have waited over 15mins and nothing. Thanks, Ryan Trauernicht __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __
Re: [OSL | CCIE_Voice] Unable to download IPMA Console
It should be 1 / http://10.22.200.21/ma/Install/IPMAConsoleInstall.jsp What SR are you on? It is SR depend for 4.13 also. Thanks, Ryan Trauernicht On Thu, Jan 15, 2009 at 4:06 PM, anil batra anil...@yahoo.com wrote: Hi All, I am trying to download IPMA Conlsoe from the follwoing URL. It shows me The page cannot be found Tried downloading on CCM server / from my laptop.. http://10.22.200.21//ma/Install/IPMAConsoleInstall.jsp Am I missing somehting here. regds // anil
Re: [OSL | CCIE_Voice] VPIM using IP Addres
What do you need help with? Kind of need to be alittle more specific. On Thu, Jan 15, 2009 at 9:05 PM, anil batra anil...@yahoo.com wrote: Hello Group, Anyone out threre please who has done CUE -Unity integration uing IP Address and can helps us..will be very kind of you. regds // anil
Re: [OSL | CCIE_Voice] Multicast MOH
can someone explain what you mean there? thanks, Ryan Trauernicht On Wed, Jan 14, 2009 at 12:39 AM, kamal yousaf lovingprin...@gmail.comwrote: I always forget g711 includes any thing g711 and below. How stupid i am. Thanks alot Vik . On Wed, Jan 14, 2009 at 5:32 PM, Vik Malhi vma...@ipexpert.com wrote: Firstly you cannot transcode a multicast stream so you are correct there. By placing the MOH server in a g711 Device Pool you are allowing all codecs that take up less bandwidth than g711 too. So that includes g729. So providing you change the IP Voice Media Streaming service params to allow G729 the MOH stream is being sent to the BR1 natively using g729. -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: *vma...@ipexpert.com * Join our free online support and peer group communities: *http://www.IPexpert.com/communities *IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- *From: *kamal yousaf lovingprin...@gmail.com *Date: *Wed, 14 Jan 2009 16:05:05 +1100 *To: *ccie_voice@onlinestudylist.com *Subject: *[OSL | CCIE_Voice] Multicast MOH Hi , I have MOH Sub and Pub configured to support Multicast for Br1 phones and Unicast for HQ phones (using MRGL).I placed MOH sub in G711 only device pool so that it communicates using G711 only. Now, since BR1phones/BR1 MGCP gw are in a device pool which communicates G729 to other device pools, how would my Multicast MOH get streamed to BR1 phones ? Multicast MOH server is using G711 , BR1 phone is using G729 and since there can be no transcoder invoked for Multicast MOH , how will this work ? Please help !
Re: [OSL | CCIE_Voice] BACD Voip peers
yup. Gatekeeper looks at g711ulaw as 2 (64k) call legs for a total of 128. Thanks, ryan Trauernicht On Wed, Jan 14, 2009 at 8:34 AM, Chris Parker cpar...@cparker.us wrote: Vik, When I type no gateway and try the call again it goes through. So I must be running into this issue. I do have bandwidth total configured on my GK as well. It is set to 96. I'm guessing if I bump it up to 128 to allow a g711 call it'll work? Chris Vik Malhi wrote: Jose is about to bring a very complicated problem with using the bandwidth total command inside gatekeeper and how it impacts B-ACD. Chris- please make the call to the B-ACD AA from a CME phone and paste the output of debug ras (assuming the router is registered to a gatekeeper). -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: _vma...@ipexpert.com _ Join our free online support and peer group communities: _http://www.IPexpert.com/communities _IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. *From: *Jose Gregorio Linero (jlinero) jlin...@cisco.com *Date: *Tue, 13 Jan 2009 22:03:59 -0500 *To: *Chris Parker cpar...@cparker.us, ccie_voice@onlinestudylist.com *Subject: *Re: [OSL | CCIE_Voice] BACD Voip peers Hi Chris: Is this router registered to a gatekeeper?. Regards, Jose -Original Message- From: Chris Parker [mailto:cpar...@cparker.us] Sent: Tuesday, January 13, 2009 09:56 PM Eastern Standard Time To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] BACD Voip peers I have had problems getting BACD to dial using voip from the phones on CME. I can dial into the BACD fine from the PSTN, but not from my IP phones. Here is my config: voice service voip allow-connections h323 to h323 dial-peer voice 3500 pots service aa incoming called-number 3500 port 0/2/0:23 ! dial-peer voice 3501 voip destination-pattern 3500 session target ipv4:172.16.101.3 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 3502 voip service aa incoming called-number 3500 dtmf-relay h245-alphanumeric codec g711ulaw no vad Eveytime I call the number I get no circuit 128A : 67 12560ms.53 +-1 +3280 pid:20004 Answer 3004 dur 00:00:00 tx:0/0 rx:0/0 22 (no circuit (34)) Telephony 50/0/4 (67) [50/0/4.0] tx:0/0/0ms None noise:0dBm acom:0dBm long duration call detected:n long dur callduration :n/a timestamp:n/a128F : 71 125594800ms.54 +-1 +30 pid:3501 Originate 3500 dur 00:00:00 tx:0/0 rx:0/0 22 (no circuit (34)) IP 0.0.0.0:0 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 pre-ietf TextRelay: off media inactive detected:n media contrl rcvd:n/a timestamp:n/a long duration call detected:n long dur callduration :n/a timestamp:n/a
[OSL | CCIE_Voice] MLP layer 2 overhead
Can anyone tell for certain if MLP with FR is 13 bytes for overhead on layer 2 or is it 13 (MLP) + 4 (FR)? Page 33 on SRND for QOS only said 13 bytes for MLP (PPP). It doesnt say it includes FR. Vik can you comment on that? You WAN video I believe said it does, but just wanting to make sure. Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] MOH Issue
If i set my MOH server to G729 for the remote branch and put a G711 file on the flash with the following commands: moh .wav multicast moh 239.1.1.3 port 16384 route X.X.X.X X.X.X.X I get dead air is that b/c the file type loaded on the flash needs to be g729? On Wed, Jan 14, 2009 at 12:11 PM, Antonio McCarver amccar...@cciequest.comwrote: Hello group, I am at the very beginning stages of my lab prep so please forgive me if this is one of those come on newbie, you should've known that questions. I have read and re-read the MOH section in the CallManager Fundamentals book, and in the CUCM 7.x SRND and I don't see where either went into detail about the different mcast addresses 239.1.1.1, .2, or .3. My question is, where can I look to read up on them and this issue? Amp Quoting Vik Malhi vma...@ipexpert.com: The two solutions work- either you place your MOH server in a g711-always DP and your should set the SRST router to use 239.1.1.1. OR...IF you did but the MOH server in a DP that uses g729 to site B (for whatever reason) then you should set the SRST router to use 239.1.1.3. The MOH file on the flash will be sent out using the same IP Address CCM is telling the phone/gateway to listen. The phone on hold is receiving RTP packets and the payload type will be g711u- however CCM ³thinks² that the MOH server back in HQ is active and the stream is g729. But I guess that¹s the whole idea of spoofing- CCM is not aware of what is going on. The codec CCM ³thinks² is being used and the actual codec are different- but that will not affect the end result. Also- while we are on the topic of sourcing music from the flash- you all should be putting in the command: no mgcp timer receive-rtcp (in the case of an MGCP gateway) -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications.
Re: [OSL | CCIE_Voice] MLP layer 2 overhead
I guess on top of that if you do MLP with LFI is that the 13 bytes or is just MLP 13bytes of overhead. If you add in LFI how much layer 2 overhead does that add? On Wed, Jan 14, 2009 at 12:13 PM, Ryan Trauernicht ryanstudyvo...@gmail.com wrote: Can anyone tell for certain if MLP with FR is 13 bytes for overhead on layer 2 or is it 13 (MLP) + 4 (FR)? Page 33 on SRND for QOS only said 13 bytes for MLP (PPP). It doesnt say it includes FR. Vik can you comment on that? You WAN video I believe said it does, but just wanting to make sure. Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] MOH Issue
home lab that I pulled the sample audio from the MOH folder. I set it to G711only and change the IP address to 239.1.1.1 and all is well. On Wed, Jan 14, 2009 at 12:44 PM, Vik Malhi vma...@ipexpert.com wrote: Can you post the output of debug ccm-m music all. Check that the MOH is being active using debug ephone moh. Dead air is better than tone. CCM thinks everything is working so the problem is lying in the spoofing part. I don't think it is anything to do with your MOH file- have you tried it with the music-on-hold.au that is provided? -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: *vma...@ipexpert.com * Join our free online support and peer group communities: *http://www.IPexpert.com/communities *IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- *From: *Ryan Trauernicht ryanstudyvo...@gmail.com *Date: *Wed, 14 Jan 2009 12:15:47 -0600 *To: *Antonio McCarver amccar...@cciequest.com *Cc: *ccie_voice@onlinestudylist.com *Subject: *Re: [OSL | CCIE_Voice] MOH Issue If i set my MOH server to G729 for the remote branch and put a G711 file on the flash with the following commands: moh .wav multicast moh 239.1.1.3 port 16384 route X.X.X.X X.X.X.X I get dead air is that b/c the file type loaded on the flash needs to be g729? On Wed, Jan 14, 2009 at 12:11 PM, Antonio McCarver amccar...@cciequest.com wrote: Hello group, I am at the very beginning stages of my lab prep so please forgive me if this is one of those come on newbie, you should've known that questions. I have read and re-read the MOH section in the CallManager Fundamentals book, and in the CUCM 7.x SRND and I don't see where either went into detail about the different mcast addresses 239.1.1.1, .2, or .3. My question is, where can I look to read up on them and this issue? Amp Quoting Vik Malhi vma...@ipexpert.com: The two solutions work- either you place your MOH server in a g711-always DP and your should set the SRST router to use 239.1.1.1. OR...IF you did but the MOH server in a DP that uses g729 to site B (for whatever reason) then you should set the SRST router to use 239.1.1.3. The MOH file on the flash will be sent out using the same IP Address CCM is telling the phone/gateway to listen. The phone on hold is receiving RTP packets and the payload type will be g711u- however CCM thinks that the MOH server back in HQ is active and the stream is g729. But I guess that's the whole idea of spoofing- CCM is not aware of what is going on. The codec CCM thinks is being used and the actual codec are different- but that will not affect the end result. Also- while we are on the topic of sourcing music from the flash- you all should be putting in the command: no mgcp timer receive-rtcp (in the case of an MGCP gateway) -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications.
Re: [OSL | CCIE_Voice] MOH Issue
My fault monday mistake! I had it based on port based and not IP based. All working now. On Wed, Jan 14, 2009 at 1:16 PM, Ryan Trauernicht ryanstudyvo...@gmail.comwrote: home lab that I pulled the sample audio from the MOH folder. I set it to G711only and change the IP address to 239.1.1.1 and all is well. On Wed, Jan 14, 2009 at 12:44 PM, Vik Malhi vma...@ipexpert.com wrote: Can you post the output of debug ccm-m music all. Check that the MOH is being active using debug ephone moh. Dead air is better than tone. CCM thinks everything is working so the problem is lying in the spoofing part. I don't think it is anything to do with your MOH file- have you tried it with the music-on-hold.au that is provided? -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: *vma...@ipexpert.com * Join our free online support and peer group communities: *http://www.IPexpert.com/communities *IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- *From: *Ryan Trauernicht ryanstudyvo...@gmail.com *Date: *Wed, 14 Jan 2009 12:15:47 -0600 *To: *Antonio McCarver amccar...@cciequest.com *Cc: *ccie_voice@onlinestudylist.com *Subject: *Re: [OSL | CCIE_Voice] MOH Issue If i set my MOH server to G729 for the remote branch and put a G711 file on the flash with the following commands: moh .wav multicast moh 239.1.1.3 port 16384 route X.X.X.X X.X.X.X I get dead air is that b/c the file type loaded on the flash needs to be g729? On Wed, Jan 14, 2009 at 12:11 PM, Antonio McCarver amccar...@cciequest.com wrote: Hello group, I am at the very beginning stages of my lab prep so please forgive me if this is one of those come on newbie, you should've known that questions. I have read and re-read the MOH section in the CallManager Fundamentals book, and in the CUCM 7.x SRND and I don't see where either went into detail about the different mcast addresses 239.1.1.1, .2, or .3. My question is, where can I look to read up on them and this issue? Amp Quoting Vik Malhi vma...@ipexpert.com: The two solutions work- either you place your MOH server in a g711-always DP and your should set the SRST router to use 239.1.1.1. OR...IF you did but the MOH server in a DP that uses g729 to site B (for whatever reason) then you should set the SRST router to use 239.1.1.3. The MOH file on the flash will be sent out using the same IP Address CCM is telling the phone/gateway to listen. The phone on hold is receiving RTP packets and the payload type will be g711u- however CCM thinks that the MOH server back in HQ is active and the stream is g729. But I guess that's the whole idea of spoofing- CCM is not aware of what is going on. The codec CCM thinks is being used and the actual codec are different- but that will not affect the end result. Also- while we are on the topic of sourcing music from the flash- you all should be putting in the command: no mgcp timer receive-rtcp (in the case of an MGCP gateway) -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications.
Re: [OSL | CCIE_Voice] Layer 2 overhead
Ok good.. FRF.12 w/ FR = 8 FR = 4 FRF.12 = 4 sorry for the confusion. On Wed, Jan 14, 2009 at 1:58 PM, wafers44 wafer...@gmail.com wrote: FR = 4 bytes FRF.12 = 8 bytes Agreed. For MLPoFR (w/ or w/out LFI - but in our case we would only be using MLP for LFI) I've been using 4B (FR) + 13B (MLP). Also, in all the IPExpert solution guides for Volume 3 atleast they've been using 4B+13B for MLPoFR On Wed, Jan 14, 2009 at 1:50 PM, Ryan Trauernicht ryanstudyvo...@gmail.com wrote: Reading the SRND and a few other books on the WAN QOS... looks like FR layer 2 is not included in the page 33 statements. Vik can you confirm these are correct for layer 2 byte sizes. MLP in the SRST states it is 13 bytes for layer 2. That actually includes LFI MLP without LFI FR = 10 bytes MLP with LFI and without FR = 13 bytes MLP with LFI FR = 17 bytes FR = 4 bytes FRF.12 = 8 bytes anyone agree or disagree? Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] Layer 2 overhead
Vik any reason why the IPExperts lab do 13+4 for MLPoFR? On Wed, Jan 14, 2009 at 2:35 PM, Shadab Abbasi (moabbasi) moabb...@cisco.com wrote: And yes, its 77 w/o compression 39 with compression Regards, Shadab CCIE# 22893 (Voice) Technology Solutions Network ~Sent from my NOKIA E61i~ -Original Message- From: anil batra [mailto:anil...@yahoo.com anil...@yahoo.com] Sent: Thursday, January 15, 2009 04:26 AM China Standard Time To: Ryan Trauernicht; ccie_voice@onlinestudylist.com; Vik Malhi Subject:Re: [OSL | CCIE_Voice] Layer 2 overhead sorry need to add RTP/UDP/Header 40 too to these vlaues..so shall it be MLP with LFI = 20+40+4+13 =77 or 20+40-+13 =73 --- On Thu, 1/15/09, anil batra anil...@yahoo.com wrote: From: anil batra anil...@yahoo.com Subject: Re: [OSL | CCIE_Voice] Layer 2 overhead To: Ryan Trauernicht ryanstudyvo...@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com, Vik Malhi vma...@ipexpert.com Date: Thursday, January 15, 2009, 1:52 AM So Vik when it says don't use FRF.12 , that is we got to use MLP LFI in that case how much the payload be 20+4+13 =17 or 20+13 = 33 Kindly let us know... --- On Thu, 1/15/09, Vik Malhi vma...@ipexpert.com wrote: From: Vik Malhi vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] Layer 2 overhead To: Ryan Trauernicht ryanstudyvo...@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Thursday, January 15, 2009, 1:42 AM MLPoFR is actually either 10 or 11 bytes (my memory is failing me). In the SRND it states MLP is 13 bytes. The 13 bytes I can only imagine is a conservative estimate or is MLPoATM. It certainly is very conservative for MLPoFR. I would clarify with the proctor- I would not use 13 + 4 = 17 bytes. Page 33 of the QoS SRND talks about these values and I would treat the 13 bytes listed for MLP as being appropriate for MLPoFR. -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Ryan Trauernicht ryanstudyvo...@gmail.com Date: Wed, 14 Jan 2009 13:50:10 -0600 To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Layer 2 overhead Reading the SRND and a few other books on the WAN QOS... looks like FR layer 2 is not included in the page 33 statements. Vik can you confirm these are correct for layer 2 byte sizes. MLP in the SRST states it is 13 bytes for layer 2. That actually includes LFI MLP without LFI FR = 10 bytes MLP with LFI and without FR = 13 bytes MLP with LFI FR = 17 bytes FR = 4 bytes FRF.12 = 8 bytes anyone agree or disagree? Thanks, Ryan Trauernicht
[OSL | CCIE_Voice] NTP CM
Any reason why CM will not keep its NTP clock. I have a local router with the following commands: ntp master 3 ntp source loopback0 (IP address is 192.168.187.1) I have edited the c:\WINNT\System32\Drivers\Etc\ntp.config My file looks like: server 192.168.187.1 # Set Local Clock to Authoritive Time Source fudge 192.168.187.1 stratum 5 # Resets Stratum from default 3 to 5 driftfile C:\WINNT\system32\drivers\etc\ntp.drift # path for drift file I stopped the Network Time Protocol service and I run ntpdate.exe 192.168.187.1 command from the cmd. That sets the clock to sync to NTP router just fine. After I reboot CM it goes back to GMT it looks like. I see it trying to sync but it never does. I have waited over 15mins and nothing. Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] NTP CM
That did not help Digging alittle bit further into this I see that my CM is actually pulling clock from my ESX box. Not sure why that is happening. Anyone else running ESX for their Call Manager having the same issue? On Wed, Jan 14, 2009 at 10:41 PM, karuna durai karu...@gmail.com wrote: Hi, After editing the ntp.conf file please fo to CMD as C:\Prog file\cisco\xntp ntpdate -b IPADD of NTP pls try this and let me know On Thu, Jan 15, 2009 at 4:33 AM, Ryan Trauernicht ryanstudyvo...@gmail.com wrote: Any reason why CM will not keep its NTP clock. I have a local router with the following commands: ntp master 3 ntp source loopback0 (IP address is 192.168.187.1) I have edited the c:\WINNT\System32\Drivers\Etc\ntp.config My file looks like: server 192.168.187.1 # Set Local Clock to Authoritive Time Source fudge 192.168.187.1 stratum 5 # Resets Stratum from default 3 to 5 driftfile C:\WINNT\system32\drivers\etc\ntp.drift # path for drift file I stopped the Network Time Protocol service and I run ntpdate.exe 192.168.187.1 command from the cmd. That sets the clock to sync to NTP router just fine. After I reboot CM it goes back to GMT it looks like. I see it trying to sync but it never does. I have waited over 15mins and nothing. Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] IPCC JTAPI Control Group Hanging
Mine I used enhance. On Tue, Jan 13, 2009 at 1:36 AM, Christian Hennrich christian.hennr...@intact-is.com wrote: I had that problem at graded labs and at home. Therefore, I have firstly not mentioned it, but I will try again configuring everything in my way on PL and give you an update. Ryan, which license version do you have installed? On Graded Labs it was the standard version. Regards ___ Christian Hennrich Senior IP Telephony Consultant INTACT integrated services GmbH Kaiserin-Augusta-Allee 113 10553 Berlin, Germany Tel: +49 30 397 35 276 Fax: +49 30 397 35 199 www.intact-is.com https://mailserver1.logicalis.de/exchweb/bin/redir.asp?URL=http://www.intact-is.com/ ___ INTACT Integrated Services GmbH eingetragen HRB 102478 beim Amtsgericht Berlin-Charlottenburg, Steuernummer 29/441/01277. Geschäftsführer: Robert Dalton, Christian Reichert From: Vik Malhi [mailto:vma...@ipexpert.com] Sent: Mon 12.01.2009 23:17 To: Ryan Trauernicht Cc: Christian Hennrich; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IPCC JTAPI Control Group Hanging Ryan/Christian- were you on one of Proctorlab's PODS or were you using your own equipment? If it was PL can you let me know which POD you tested onI'll try to get to the bottom of this. -- Vik Malhi - CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Ryan Trauernicht ryanstudyvo...@gmail.com Date: Mon, 12 Jan 2009 15:16:57 -0600 To: Vik Malhi vma...@ipexpert.com Cc: Christian Hennrich christian.hennr...@intact-is.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IPCC JTAPI Control Group Hanging Correct. In IPCC for the JTAPI Provider put Sub,Pub (192.168.187.12,192.168.187.11) Then try to add in a group. It doesnt matter what DP I pick. If I change the Provider to only the Pub and you are good to go. Also restarting the node manager. Thanks, Ryan Trauernicht On Mon, Jan 12, 2009 at 3:09 PM, Vik Malhi vma...@ipexpert.com wrote: I'm renaming Default to HQ. When you say you are only using the PUB and not the SUB do you mean that when you define the JTAPI Provider you are only specifying the PUB and not the SUB? -- Vik Malhi - CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Christian Hennrich christian.hennr...@intact-is.com Date: Mon, 12 Jan 2009 22:03:37 +0100 To: Vik Malhi vma...@ipexpert.com Cc: Ryan Trauernicht ryanstudyvo...@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IPCC JTAPI Control Group Hanging I have to 101% to agree with Ryan, that I ran in the same problem, if __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __
Re: [OSL | CCIE_Voice] MOH Issue
If you want to play a G711 file (which is required) on a G729 stream you need to change your multicast address to +2... 239.1.1.1 (base in CM) you would need 239.1.1.3 on the router. Also dont forget the ccm-manager music-on-hold command. Thanks, Ryan Trauernicht 2009/1/13 saralilin2...@yahoo.co.jp create a g711 only region put moh in it. moh to sb will be g711 *Kumar, Narinder narinder.ku...@uxcg.com.au* wrote: The file in flash is g711 moh SampleAudioSource.ULAW.wav *From:* saralilin2...@yahoo.co.jp [mailto:saralilin2...@yahoo.co.jp] *Sent:* Tuesday, 13 January 2009 11:44 PM *To:* Kumar, Narinder; ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] MOH Issue flash moh only play g711 not g729 *Kumar, Narinder narinder.ku...@uxcg.com.au* wrote: Quick Que on MOH CCM running multicast MOH. Between Site A and Site B only g729 allowed SiteA will receive multicast MOH . Site B will receive multicast MOH from router flash, no multicast traffic allowed between Ste A and SiteB. The way I do this question is Configure the MOH source file and tick multicast and play continuously Enable multicast on the MRG and MOH server Change the ip voice media service parameter to allow both g711 and g729 Site A works without any issue Site B Configuration: Call-manager-fallback Moh filename ( Moh file in flash) multicast moh 239.1.1.1 port 16384 route x.x.x.x MOH from site B doesn't work , what am I missing here ? *** debug ccm-manager music-on-hold all ** an 13 13:13:30.023: moh_update_rtp: callID 17 dstCallID 18 *Jan 13 13:13:30.023: moh_process_ccb: dstadr 0.0.0.0, callid 18, port 0, codec 65535, moh_en 0, moh_addr 0.0.0.0 *Jan 13 13:13:30.023: moh_update_rtp: callID 17 dstCallID 18 *Jan 13 13:13:30.079: moh_process_ccb: dstadr 142.102.65.6, callid 18, port 23552, codec 5, moh_en 0, moh_addr 0.0.0.0 *Jan 13 13:13:30.079: moh_update_rtp: callID 17 dstCallID 18 *Jan 13 13:13:31.391: %ISDN-6-CONNECT: Interface Serial0/1/0:2 is now connected to 911 N/A *Jan 13 13:13:31.395: moh_update_rtp: callID 17 dstCallID 18 *Jan 13 13:13:31.395: moh_process_ccb: dstadr 142.102.65.6, callid 18, port 23552, codec 5, moh_en 0, moh_addr 0.0.0.0 *Jan 13 13:13:31.399: moh_update_rtp: callID 17 dstCallID 18 *Jan 13 13:13:33.103: moh_update_rtp: callID 17 dstCallID 18 *Jan 13 13:13:33.103: moh_update_rtp: callID 17 dstCallID 18 *Jan 13 13:13:33.103: moh_update_rtp: callID 17 dstCallID 18 *Jan 13 13:13:33.119: moh_update_rtp: callID 17 dstCallID 18 *Jan 13 13:13:33.139: moh_process_ccb: dstadr 239.1.1.3, callid 18, port 16384, codec 16, moh_en 0, moh_addr 0.0.0.0 *Jan 13 13:13:33.139: moh_process_ccb:multicast addr add_ccb *Jan 13 13:13:33.139: moh_add_ccb: ip addr 239.1.1.3 port 16384 callid 18 *Jan 13 13:13:33.139: moh_add_ccb: vmccb does not exists - creating a new one for 239.1.1.3 through IGMP *Jan 13 13:13:33.139: moh_join_group_command called for 239.1.1.3 *Jan 13 13:13:33.139: moh_join_group_command: Looking at valid idb's to configure 239.1.1.3 *Jan 13 13:13:33.139: moh_join_group_command: IGMP API on group 239.1.1.3 idb Se0/0/0.201 *Jan 13 13:13:33.139: moh_join_group_command: IGMP API on group 239.1.1.3 idb Vl102 *Jan 13 13:13:33.139: moh_create_session: called *Jan 13 13:13:33.139: moh_create_session : dstadr 239.1.1.3 does not exist - creating acontrol block *Jan 13 13:13:33.139: moh_insert_multicast_hashtable:moh_insert_multicast_hashtable buc 2 *Jan 13 13:13:33.139: moh_create_session : Created a new vmccb for 239.1.1.3 *Jan 13 13:13:33.139: moh_send_join: Looking at valid idb's to configure 239.1.1.3 *Jan 13 13:13:33.139: moh_add_ccb: Done inserting CCB for 239.1.1.3 *Jan 13 13:13:33.139: moh_update_rtp: callID 17 dstCallID 18 *Jan 13 13:13:36.091: moh_update_rtp: callID 17 dstCallID 18 *Jan 13 13:13:36.091: moh_update_rtp: callID 17 dstCallID 18 *Jan 13 13:13:36.091: update_stream_info: stream_flag Reset *Jan 13 13:13:36.091: moh_update_rtp: callID 17 dstCallID 18 *Jan 13 13:13:36.091: update_stream_info: stream_flag Reset *Jan 13 13:13:36.115: moh_update_rtp: callID 17 dstCallID 18 *Jan 13 13:13:36.115: update_stream_info: stream_flag Reset *Jan 13 13:13:36.183: moh_process_ccb: dstadr 142.102.65.6, callid 18, port 24164, codec 5, moh_en 1, moh_addr 239.1.1.3 *Jan 13 13:13:36.183: moh_process_ccb:multicast addr delete_ccb call id 18 moh_call_id 18 *Jan 13 13:13:36.183: moh_delete_ccb: called dstadr 239.1.1.3, callid 18 *Jan 13 13:13:36.183: moh_delete_ccb_ext:called dstadr 239.1.1.3, callid 18 *Jan 13 13:13:36.183: moh_delete_ccb_ext:ipaddr 239.1.1.3 callid 18 *Jan 13 13:13:36.183: moh_delete_ccb_ext
Re: [OSL | CCIE_Voice] 3rd Day in raw :( SRST-unity integration problem , even CTI solution doesn't work,
Is this proctor labs or your own personal lab? On Tue, Jan 13, 2009 at 12:12 PM, jeremy co jeremy.coo...@gmail.com wrote: Hi, I tried both DRNIS and CTI solution. None of them worked .in both ,8 digits passed to unity, I donnow why!!! ** RDNIS scenario : unity--HQ ---pstn-BR1 (SRST) 2001 3001 2001 call 3001 and CFNA redirect call to unity via pstn , redirecting number works fine but only 8 digits passed to unity Here is the out put of debug isdn on HQ when call forwarded to unity. HQ :499-202-2 BR1 :899-303-3XXX voice pilot number : 2229 Mar 11 20:01:40.060: ISDN Se0/0:23 Q931: RX - SETUP pd = 8 callref = 0x008E Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Calling Party Number i = 0x2181, '4992022002' Plan:ISDN, Type:National Called Party Number i = 0xA1, '499209' Plan:ISDN, Type:National Redirecting Number i = 0xFF, '8993033001' Plan:Reserved, Type:Reserved I can see from call viewer in unity : dialed numbercalling number forwarding 93033001 4992022002 93033001 CTI scenraio : I made 2888 CTI with forward to voice mail option checked,and assign Voice mail profile with mask of to it. Then Configured SRST to forward all calls to 2888 DN. what I see in unity is again dialed numbercalling number forwarding 93033001 4992022002 93033001 I have no idea why 8 digits just passed to unity I waste lots of time to make SRST to work, but no success any help would be much appreciated. Jeremy
Re: [OSL | CCIE_Voice] BACD Voip peers
I always combine the voips.. dial-peer voice 3502 voip service aa destination-pattern 3500 incoming called-number 3500 dtmf-relay h245-alphanumeric codec g711ulaw no vad ip qos dscp cs3 sign then prob need to do a call application voice load aa call application voice load acd What does your BACD commands look like and what are you trying to accomplist... drop through or _welcome_prompt? Thanks, Ryan Trauernicht On Tue, Jan 13, 2009 at 8:55 PM, Chris Parker cpar...@cparker.us wrote: I have had problems getting BACD to dial using voip from the phones on CME. I can dial into the BACD fine from the PSTN, but not from my IP phones. Here is my config: voice service voip allow-connections h323 to h323 dial-peer voice 3500 pots service aa incoming called-number 3500 port 0/2/0:23 ! dial-peer voice 3501 voip destination-pattern 3500 session target ipv4:172.16.101.3 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 3502 voip service aa incoming called-number 3500 dtmf-relay h245-alphanumeric codec g711ulaw no vad Eveytime I call the number I get no circuit 128A : 67 12560ms.53 +-1 +3280 pid:20004 Answer 3004 dur 00:00:00 tx:0/0 rx:0/0 22 (no circuit (34)) Telephony 50/0/4 (67) [50/0/4.0] tx:0/0/0ms None noise:0dBm acom:0dBm long duration call detected:n long dur callduration :n/a timestamp:n/a128F : 71 125594800ms.54 +-1 +30 pid:3501 Originate 3500 dur 00:00:00 tx:0/0 rx:0/0 22 (no circuit (34)) IP 0.0.0.0:0 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 pre-ietf TextRelay: off media inactive detected:n media contrl rcvd:n/a timestamp:n/a long duration call detected:n long dur callduration :n/a timestamp:n/a
Re: [OSL | CCIE_Voice] BACD Voip peers
Good point Jose If the CME is registered to the GK and you have BW restrictions to 32k or something lower then 128 the call will never complete. Even though the voip dial peer is not a ras it will still as the GK for bw limitations. Thanks, Ryan Trauernicht On Tue, Jan 13, 2009 at 9:05 PM, Ryan Trauernicht ryanstudyvo...@gmail.comwrote: I always combine the voips.. dial-peer voice 3502 voip service aa destination-pattern 3500 incoming called-number 3500 dtmf-relay h245-alphanumeric codec g711ulaw no vad ip qos dscp cs3 sign then prob need to do a call application voice load aa call application voice load acd What does your BACD commands look like and what are you trying to accomplist... drop through or _welcome_prompt? Thanks, Ryan Trauernicht On Tue, Jan 13, 2009 at 8:55 PM, Chris Parker cpar...@cparker.us wrote: I have had problems getting BACD to dial using voip from the phones on CME. I can dial into the BACD fine from the PSTN, but not from my IP phones. Here is my config: voice service voip allow-connections h323 to h323 dial-peer voice 3500 pots service aa incoming called-number 3500 port 0/2/0:23 ! dial-peer voice 3501 voip destination-pattern 3500 session target ipv4:172.16.101.3 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 3502 voip service aa incoming called-number 3500 dtmf-relay h245-alphanumeric codec g711ulaw no vad Eveytime I call the number I get no circuit 128A : 67 12560ms.53 +-1 +3280 pid:20004 Answer 3004 dur 00:00:00 tx:0/0 rx:0/0 22 (no circuit (34)) Telephony 50/0/4 (67) [50/0/4.0] tx:0/0/0ms None noise:0dBm acom:0dBm long duration call detected:n long dur callduration :n/a timestamp:n/a128F : 71 125594800ms.54 +-1 +30 pid:3501 Originate 3500 dur 00:00:00 tx:0/0 rx:0/0 22 (no circuit (34)) IP 0.0.0.0:0 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 pre-ietf TextRelay: off media inactive detected:n media contrl rcvd:n/a timestamp:n/a long duration call detected:n long dur callduration :n/a timestamp:n/a
Re: [OSL | CCIE_Voice] unity port allocation to MWI dialout role. Please Vic comment on this.
Jeremy, The would tell you in the lab what to do. If they say to configure 3 Unity ports and don't give you any more specific details... assume that all 3 ports can do all rolls. answer calls, TRAP, MWI notification and notification dialout. thanks, Ryan Trauernicht On Mon, Jan 12, 2009 at 3:21 AM, jeremy co jeremy.coo...@gmail.com wrote: Hi, I need clarification on this. If question ask to configure say 3 ports on unity. there two options: - 3ports both Answering calls and MWI dialout -2 ports Answering calls and 1 port MWI dialout SO which approch is correct, I think for real world we use first option, but for perspective of lab ,which one is correct? Jeremy
[OSL | CCIE_Voice] Thoughts on SIP Trunk with multicast MOH
Anyway around getting multicast MOH to flow over to a SIP trunk? Anytime I hit the hold and unhold the cool just hangs. Change it to unicast and it works fine. This is not a MTP or xcoding issue. I see the MTP invoked on a g711only call. Any way around if they require multicast at HQ and getting supplementary features to work? thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] IPCC JTAPI Control Group Hanging
Yup that is exactly what I do. In CM i have 4 defined DP (HQ, SiteB, 711, 729). In IPCC I see 5 (adding in Default). I pick the HQ DP for the adding in JTAPI and it just hangs. Only way I can get around it was making the JTAPI integration just point to publisher. On Mon, Jan 12, 2009 at 1:08 PM, Vik Malhi vma...@ipexpert.com wrote: When you use Default DP it literally looks for a DP called Default and will not use the default defined in Device Defaults with CCM. Try saying the above statement 10 times! If you rename the Default DP to HQ then when adding the JTAPI Call Control Group it will hang. Do not do this- select the HQ DP. -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: *vma...@ipexpert.com * Join our free online support and peer group communities: *http://www.IPexpert.com/communities *IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- *From: *Ryan Trauernicht ryanstudyvo...@gmail.com *Date: *Sun, 11 Jan 2009 16:46:39 -0600 *To: *ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com *Subject: *[OSL | CCIE_Voice] IPCC JTAPI Control Group Hanging What is the work around for IPCC express when trying to add in a JTAPI controlled group with the default DP? It just sits there and does nothing. Only way I found around it is to set the JTAPI integration only to the publisher and configure everything... then add in the sub later. Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] IPCC JTAPI Control Group Hanging
Vik, Put in the Sub,Pub (not Pub,Sub). On Mon, Jan 12, 2009 at 3:03 PM, Christian Hennrich christian.hennr...@intact-is.com wrote: Vik, Ryan, I have to 101% to agree with Ryan, that I ran in the same problem, if I'm using pub and sub. Only using pub works. I'm renaming the defautl DP to HQ and use that for the CTI Ports. VIK do you rename the default DP or are you creating a new one? Regards Vik Malhi schrieb: Hmmm...I'm not running into that myself. I define the JTAPI provider like so: 10.x.200.21,10.x.200.20 (PUB,SUB). Then ensure CTI Manager is running on both. I've never seen what you are describing before:-( -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: _vma...@ipexpert.com _ Join our free online support and peer group communities: _http://www.IPexpert.com/communities _IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. *From: *Ryan Trauernicht ryanstudyvo...@gmail.com *Date: *Mon, 12 Jan 2009 13:30:19 -0600 *To: *Vik Malhi vma...@ipexpert.com *Cc: *ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com *Subject: *Re: [OSL | CCIE_Voice] IPCC JTAPI Control Group Hanging Yup that is exactly what I do. In CM i have 4 defined DP (HQ, SiteB, 711, 729). In IPCC I see 5 (adding in Default). I pick the HQ DP for the adding in JTAPI and it just hangs. Only way I can get around it was making the JTAPI integration just point to publisher. On Mon, Jan 12, 2009 at 1:08 PM, Vik Malhi vma...@ipexpert.com wrote: When you use Default DP it literally looks for a DP called Default and will not use the default defined in Device Defaults with CCM. Try saying the above statement 10 times! If you rename the Default DP to HQ then when adding the JTAPI Call Control Group it will hang. Do not do this- select the HQ DP. --Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: _vma...@ipexpert.com http://vma...@ipexpert.com _ Join our free online support and peer group communities: _http://www.IPexpert.com/communities _IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. *From: *Ryan Trauernicht ryanstudyvo...@gmail.com http://ryanstudyvo...@gmail.com *Date: *Sun, 11 Jan 2009 16:46:39 -0600 *To: *ccie_voice@onlinestudylist.com http://ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com http://ccie_voice@onlinestudylist.com *Subject: *[OSL | CCIE_Voice] IPCC JTAPI Control Group Hanging What is the work around for IPCC express when trying to add in a JTAPI controlled group with the default DP? It just sits there and does nothing. Only way I found around it is to set the JTAPI integration only to the publisher and configure everything... then add in the sub later. Thanks, Ryan Trauernicht __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __
Re: [OSL | CCIE_Voice] IPCC JTAPI Control Group Hanging
Correct. In IPCC for the JTAPI Provider put Sub,Pub (192.168.187.12,192.168.187.11) Then try to add in a group. It doesnt matter what DP I pick. If I change the Provider to only the Pub and you are good to go. Also restarting the node manager. Thanks, Ryan Trauernicht On Mon, Jan 12, 2009 at 3:09 PM, Vik Malhi vma...@ipexpert.com wrote: I'm renaming Default to HQ. When you say you are only using the PUB and not the SUB do you mean that when you define the JTAPI Provider you are only specifying the PUB and not the SUB? -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Christian Hennrich christian.hennr...@intact-is.com Date: Mon, 12 Jan 2009 22:03:37 +0100 To: Vik Malhi vma...@ipexpert.com Cc: Ryan Trauernicht ryanstudyvo...@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IPCC JTAPI Control Group Hanging I have to 101% to agree with Ryan, that I ran in the same problem, if
Re: [OSL | CCIE_Voice] IPCC JTAPI Control Group Hanging
I am using my home lab. Not the PL. Sorry! On Mon, Jan 12, 2009 at 4:17 PM, Vik Malhi vma...@ipexpert.com wrote: Ryan/Christian- were you on one of Proctorlab's PODS or were you using your own equipment? If it was PL can you let me know which POD you tested onI'll try to get to the bottom of this. -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: *vma...@ipexpert.com * Join our free online support and peer group communities: *http://www.IPexpert.com/communities *IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- *From: *Ryan Trauernicht ryanstudyvo...@gmail.com *Date: *Mon, 12 Jan 2009 15:16:57 -0600 *To: *Vik Malhi vma...@ipexpert.com *Cc: *Christian Hennrich christian.hennr...@intact-is.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com *Subject: *Re: [OSL | CCIE_Voice] IPCC JTAPI Control Group Hanging Correct. In IPCC for the JTAPI Provider put Sub,Pub (192.168.187.12,192.168.187.11) Then try to add in a group. It doesnt matter what DP I pick. If I change the Provider to only the Pub and you are good to go. Also restarting the node manager. Thanks, Ryan Trauernicht On Mon, Jan 12, 2009 at 3:09 PM, Vik Malhi vma...@ipexpert.com wrote: I'm renaming Default to HQ. When you say you are only using the PUB and not the SUB do you mean that when you define the JTAPI Provider you are only specifying the PUB and not the SUB? -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Christian Hennrich christian.hennr...@intact-is.com Date: Mon, 12 Jan 2009 22:03:37 +0100 To: Vik Malhi vma...@ipexpert.com Cc: Ryan Trauernicht ryanstudyvo...@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IPCC JTAPI Control Group Hanging I have to 101% to agree with Ryan, that I ran in the same problem, if
Re: [OSL | CCIE_Voice] IPCC Transfer Option While Queued w/ Hold Music
What prompt do you choose to play when configuring the timeout to 30? IF you dont pick a valid prompt doesnt it goto unsuccessful? On Sun, Jan 11, 2009 at 4:09 PM, Ryan Trauernicht ryanstudyvo...@gmail.comwrote: That options seems like it would save alot of time. I have always just pulled the MOH file from CM and used sound recorder to corp it to the delay length. Then I play that prompt in a menu statement with option 0. So the prompt is actually playing the MOH file. No need for hold/menu/unhold On Fri, Jan 9, 2009 at 11:41 PM, James Oberhaus joberh...@gmail.comwrote: Thanks Vik and Scott... using the timeout in the Menu option instead of Delay was the key! On Fri, Jan 9, 2009 at 8:40 PM, ccie_voice-requ...@onlinestudylist.comwrote: Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: CM call to CME Phone and CME puts CM on park... call disconnects (Vik Malhi) 2. Re: Device Pool Question (Vik Malhi) 3. QOS Marking (Hany Hanna) 4. Ephones still registering with gatekeeper withno-reg!! (jeremy co) 5. ATA conversion from SCCP to SIP (Scott ODonnell) 6. Re: IPCC Transfer Option While Queued w/ Hold Music (Scott ODonnell) -- Message: 1 Date: Fri, 09 Jan 2009 19:14:57 -0800 From: Vik Malhi vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] CM call to CME Phone and CME puts CM on park... call disconnects To: Ryan Trauernicht ryanstudyvo...@gmail.com, anil...@yahoo.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Message-ID: c58d52b1.4ac8%vma...@ipexpert.comc58d52b1.4ac8%25vma...@ipexpert.com Content-Type: text/plain; charset=iso-8859-1 Wait for H245 TCS on the trunk page within CCMAdmin should be unchecked. Let us know how you get on with that. -- Vik Malhi ? CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Ryan Trauernicht ryanstudyvo...@gmail.com Date: Fri, 9 Jan 2009 18:58:45 -0600 To: anil...@yahoo.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CM call to CME Phone and CME puts CM on park... call disconnects Should I be checking the MTP box on the GK (225 controlled) trunk? I didnt think I was suppose to have that checked. On Fri, Jan 9, 2009 at 6:21 PM, anil batra anil...@yahoo.com wrote: wt's ur MTP setting on GK. Try enabing it. --- On Sat, 1/10/09, Ryan Trauernicht ryanstudyvo...@gmail.com wrote: From: Ryan Trauernicht ryanstudyvo...@gmail.com Subject: [OSL | CCIE_Voice] CM call to CME Phone and CME puts CM on park... call disconnects To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Saturday, January 10, 2009, 5:30 AM Setup CME registered to GK to CM. Call from the PSTN to CME phone and is placed on park works fine. When a CM phone calls the CME phone and the CME phone places the CM phones on hold... disconnect. CMGKCME Phone is G729. I have a Xcoder registered and working. CM phone to CME phone and to VM works fine. Any ideas? Thanks, Ryan Trauernicht -- next part -- An HTML attachment was scrubbed... URL: http://onlinestudylist.com/pipermail/ccie_voice/attachments/20090109/ecad6624/attachment-0001.htm -- Message: 2 Date: Fri, 09 Jan 2009 19:30:54 -0800 From: Vik Malhi vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] Device Pool Question To: Sergio Polizer spoli...@hotmail.com, ccielab...@gmail.com, Mark Snow ms...@ipexpert.com Cc: ccie_voice@onlinestudylist.com Message-ID: c58d566e.4aca%vma...@ipexpert.comc58d566e.4aca%25vma...@ipexpert.com Content-Type: text/plain; charset=iso-8859-1 If you are using g729 then the SW MTP is no good since no Low Bit Rate codec is supported using the SW MTP. What is prob happening is the HQ/BR1 Xcoder is being invoked when you switch the SIP trunk DP. -- Vik Malhi ? CCIE #13890, CCSI #31584 Senior Technical Instructor
[OSL | CCIE_Voice] QOS MLP 1 link and FRF Shaping the other verification
I know when you put the frame-relay traffic-shaping on the physical interface it turns all the CIRs down to 56k. If you have 1 pipe that is MLP FRF which you need to put that command on the interface and the other pip is just shaping FRF. I just wanted to make sure you can not do the nested policy-map way. You must do it the old school map-class way for (cir, mincir, bc, be). Vik can you comment on that or anyone else who knows for sure. Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] QOS MLP 1 link and FRF Shaping the other verification
Another question is. If you use the traditional way for low speed links... (cir, mincir, bc, be) it is required to do the frame-relay traffic-shaping on the physical isnt it? On Mon, Jan 12, 2009 at 4:49 PM, Ryan Trauernicht ryanstudyvo...@gmail.comwrote: That is what I thought. do you think it is over kill if they ask you to not mark anything on the switch level but configure MLP on 1 link and FRF.12 on other other pipe to mark on the ingress of the FA and match on the outgress of the Serial? Even though I have changed the CTI, IPVMSA paramaters to mark CS3 already. I saw last night that the 6608 keepalives all still use AF31. something like this.. class-map match-any RTP-WAN match ip dscp ef class-map match-any SCCP-WAN match ip dscp cs3 class-map match-any RTP-LAN match ip dscp ef match access-group RTP class-map match-any SCCP-LAN match ip dscp cs3 match ip dscp af31 match ip access-group SCCP access-list extended RTP permit ip udp any any range 16384 32767 access-list extended SCCP permit ip tcp any any range 2000 2002 permit ip tcp any any range 11000 11999 permit ip udp any any eq 2427 permit ip tcp any any eq 2428 permit ip udp any any eq 5060 permit ip tcp any any eq 5060 permit ip tcp any any eq 1718 permit ip udp any any eq 1719 permit ip tcp any any eq 1720 On Mon, Jan 12, 2009 at 4:46 PM, Vik Malhi vma...@ipexpert.com wrote: You must do it the old school way if you are using a single physical interface. The old school way being FRTS as opposed to class-based shaping. Vik Malhi - CCIE#13890 Senior Technical Instructor - IPexpert Inc Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join IPexpert's Free CCIE Peer Groups Study Communities at www.IPexpert.com/communities On Jan 12, 2009, at 2:43 PM, Ryan Trauernicht ryanstudyvo...@gmail.com wrote: I know when you put the frame-relay traffic-shaping on the physical interface it turns all the CIRs down to 56k. If you have 1 pipe that is MLP FRF which you need to put that command on the interface and the other pip is just shaping FRF. I just wanted to make sure you can not do the nested policy-map way. You must do it the old school map-class way for (cir, mincir, bc, be). Vik can you comment on that or anyone else who knows for sure. Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] QOS MLP 1 link and FRF Shaping the other verification
Just looking to see if this is overkill for a CB Shaping method and marking (leaving everything alone on the layer 2switches). Also 5 percent (not using percent) for SCCP and 4 FRF.12 calls for priority. Is it really necessary to mark on the ingress or is is ok to trust what the phones and CM have marked as packets (as long as you adjust the service parameters). LAN QOS class-map match-any RTP-LAN match ip dscp ef match access-group name RTP class-map match-any SCCP-LAN match ip dscp cs3 match ip dscp af31 match access-group name SCCP policy-map QOS-LAN class RTP-LAN set ip dscp ef class SCCP-LAN set ip dscp cs3 ip access-list extended RTP permit udp any any range 16384 32767 ip access-list extended SCCP permit tcp any any range 2000 2002 permit tcp any any range 11000 11999 permit tcp any any eq 5060 permit udp any any eq 5060 permit tcp any any eq 1718 permit udp any any eq 1719 permit tcp any any eq 1720 permit udp any any eq 2427 permit tcp any any eq 2428 int fa0/0 service-policy input QOS-LAN WAN QOS class-map match-any RTP-WAN match ip dscp ef class-map match-any SCCP-WAN match ip dscp cs3 policy-map QOS-WAN-SiteC class RTP-WAN priority 112 class SCCP-WAN bandwidth 37 class class-default fair-queue policy-map MQC-FRTS-SiteC class class-default shape average 729600 7296 0 service-policy QOS-WAN-SiteC map-class frame-relay FR-toSiteC frame-relay fragment 960 service-policy output MQC-FRTS-SiteC int s0/0 encap frame-relay frame-relay lmi-type ansi int s0/0.100 frame-relay int-dlci class FR-toSiteC On Mon, Jan 12, 2009 at 5:18 PM, Kumar, Narinder narinder.ku...@uxcg.com.au wrote: Yes you do need to have traffic-shaping on the physical *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Ryan Trauernicht *Sent:* Tuesday, 13 January 2009 10:05 AM *To:* Vik Malhi *Cc:* ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] QOS MLP 1 link and FRF Shaping the other verification Another question is. If you use the traditional way for low speed links... (cir, mincir, bc, be) it is required to do the frame-relay traffic-shaping on the physical isnt it? On Mon, Jan 12, 2009 at 4:49 PM, Ryan Trauernicht ryanstudyvo...@gmail.com wrote: That is what I thought. do you think it is over kill if they ask you to not mark anything on the switch level but configure MLP on 1 link and FRF.12 on other other pipe to mark on the ingress of the FA and match on the outgress of the Serial? Even though I have changed the CTI, IPVMSA paramaters to mark CS3 already. I saw last night that the 6608 keepalives all still use AF31. something like this.. class-map match-any RTP-WAN match ip dscp ef class-map match-any SCCP-WAN match ip dscp cs3 class-map match-any RTP-LAN match ip dscp ef match access-group RTP class-map match-any SCCP-LAN match ip dscp cs3 match ip dscp af31 match ip access-group SCCP access-list extended RTP permit ip udp any any range 16384 32767 access-list extended SCCP permit ip tcp any any range 2000 2002 permit ip tcp any any range 11000 11999 permit ip udp any any eq 2427 permit ip tcp any any eq 2428 permit ip udp any any eq 5060 permit ip tcp any any eq 5060 permit ip tcp any any eq 1718 permit ip udp any any eq 1719 permit ip tcp any any eq 1720 On Mon, Jan 12, 2009 at 4:46 PM, Vik Malhi vma...@ipexpert.com wrote: You must do it the old school way if you are using a single physical interface. The old school way being FRTS as opposed to class-based shaping. Vik Malhi - CCIE#13890 Senior Technical Instructor - IPexpert Inc Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join IPexpert's Free CCIE Peer Groups Study Communities at www.IPexpert.com/communities On Jan 12, 2009, at 2:43 PM, Ryan Trauernicht ryanstudyvo...@gmail.com wrote: I know when you put the frame-relay traffic-shaping on the physical interface it turns all the CIRs down to 56k. If you have 1 pipe that is MLP FRF which you need to put that command on the interface and the other pip is just shaping FRF. I just wanted to make sure you can not do the nested policy-map way. You must do it the old school map-class way for (cir, mincir, bc, be). Vik can you comment on that or anyone else who knows for sure. Thanks, Ryan Trauernicht -- CONFIDENTIALITY - The information contained in this electronic mail message is confidential and is intended solely for the addressee(s). If you are not an authorised recipient of this message please contact Getronics Australia immediately by reply email and destroy/delete this message from your computer. Any unauthorised form of reproduction of this message, or part thereof, is strictly prohibited. DISCLAIMER - Unless specifically indicated otherwise, the views
Re: [OSL | CCIE_Voice] QOS MLP 1 link and FRF Shaping the other verification
You need Frame-relay Traffic-shaping command placed on the physical interface (wha tI mean by that is Serial0/0... NOT s0/0.101) So not the PVC when you use FRF.12 traditional way (aka cir, mincir, bc, be) and MLP. You do not put in on the physical interface when you use the CB Shaping way for FRF.12 (aka nested policy-maps) Thanks, Ryan Trauernicht On Mon, Jan 12, 2009 at 5:49 PM, anil batra anil...@yahoo.com wrote: 1. Let's say BR1 to HQ we are to use MLP with LFI but for BR2 to HQ we are to use FRF.12 Fragmentation. What I understand is we will use MLP with LFI between HQ-BR1 with NO frame-realy traffic-shaping command on major interface. Now on BR2 to HQ as we are supposed to use FRF.12 , in this case we will have to use Legacy FRTS for this link but not MQC-FRTS right ??? 2. I am little confused whe do you need to put frame-realy traffic-shaping command on major interface - MLP - I think NO Legacy FRTS - I think NO MQC-FRTS - I think YES regards // anil --- On *Tue, 1/13/09, Vik Malhi vma...@ipexpert.com* wrote: From: Vik Malhi vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] QOS MLP 1 link and FRF Shaping the other verification To: Ryan Trauernicht ryanstudyvo...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Tuesday, January 13, 2009, 4:16 AM You must do it the old school way if you are using a single physical interface. The old school way being FRTS as opposed to class-based shaping. Vik Malhi - CCIE#13890 Senior Technical Instructor - IPexpert Inc Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join IPexpert's Free CCIE Peer Groups Study Communities atwww.IPexpert.com/communities On Jan 12, 2009, at 2:43 PM, Ryan Trauernicht ryanstudyvo...@gmail.com wrote: I know when you put the frame-relay traffic-shaping on the physical interface it turns all the CIRs down to 56k. If you have 1 pipe that is MLP FRF which you need to put that command on the interface and the other pip is just shaping FRF. I just wanted to make sure you can not do the nested policy-map way. You must do it the old school map-class way for (cir, mincir, bc, be). Vik can you comment on that or anyone else who knows for sure. Thanks, Ryan Trauernicht
[OSL | CCIE_Voice] H323 Gateway as SiteB calling CUE (SiteC)
Do you need fast start enabled if you have an H323 Gateway that calls into CUE? Thanks, ryan Trauernicht
Re: [OSL | CCIE_Voice] QOS MLP 1 link and FRF Shaping the other verification
You will not be asked that. That can not be done. Option 2 needs to be configured as traditional method. thanks, Ryan Trauernicht On Mon, Jan 12, 2009 at 6:17 PM, anil batra anil...@yahoo.com wrote: What if we are to configure in the scenario I mentioned where it says configure 1. HQ to BR1 we are to use MLP with LFI 2. HQ to BR2 we are to use FRF.12 with MQC-FRTS (CB-Shapping way) In the above scenario, on HQ major( Physica) interface is same. But as you mentioned we should not apply Frame-relay Traffic-shaping command for MQC-FRTS but we will have to apply for MLP. In another words the above scenario should be avoided and we shoudl use Leagacy FRTS only for HQ to BR2. That means MLP and MQC sharing same physical interface are mutually exclusive. And hence we shoufl use HQ to BR1 we are to use MLP with LFI and Leagcy FRTS for HQ to BR2 we are to use FRF.12 . -anil --- On *Tue, 1/13/09, Ryan Trauernicht ryanstudyvo...@gmail.com* wrote: From: Ryan Trauernicht ryanstudyvo...@gmail.com Subject: Re: [OSL | CCIE_Voice] QOS MLP 1 link and FRF Shaping the other verification To: anil...@yahoo.com Cc: Vik Malhi vma...@ipexpert.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Tuesday, January 13, 2009, 5:26 AM You need Frame-relay Traffic-shaping command placed on the physical interface (wha tI mean by that is Serial0/0... NOT s0/0.101) So not the PVC when you use FRF.12 traditional way (aka cir, mincir, bc, be) and MLP. You do not put in on the physical interface when you use the CB Shaping way for FRF.12 (aka nested policy-maps) Thanks, Ryan Trauernicht On Mon, Jan 12, 2009 at 5:49 PM, anil batra anil...@yahoo.com wrote: 1. Let's say BR1 to HQ we are to use MLP with LFI but for BR2 to HQ we are to use FRF.12 Fragmentation. What I understand is we will use MLP with LFI between HQ-BR1 with NO frame-realy traffic-shaping command on major interface. Now on BR2 to HQ as we are supposed to use FRF.12 , in this case we will have to use Legacy FRTS for this link but not MQC-FRTS right ??? 2. I am little confused whe do you need to put frame-realy traffic-shaping command on major interface - MLP - I think NO Legacy FRTS - I think NO MQC-FRTS - I think YES regards // anil --- On *Tue, 1/13/09, Vik Malhi vma...@ipexpert.com* wrote: From: Vik Malhi vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] QOS MLP 1 link and FRF Shaping the other verification To: Ryan Trauernicht ryanstudyvo...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Tuesday, January 13, 2009, 4:16 AM You must do it the old school way if you are using a single physical interface. The old school way being FRTS as opposed to class-based shaping. Vik Malhi - CCIE#13890 Senior Technical Instructor - IPexpert Inc Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join IPexpert's Free CCIE Peer Groups Study Communities atwww.IPexpert.com/communities http://www.ipexpert.com/communities On Jan 12, 2009, at 2:43 PM, Ryan Trauernicht ryanstudyvo...@gmail.com wrote: I know when you put the frame-relay traffic-shaping on the physical interface it turns all the CIRs down to 56k. If you have 1 pipe that is MLP FRF which you need to put that command on the interface and the other pip is just shaping FRF. I just wanted to make sure you can not do the nested policy-map way. You must do it the old school map-class way for (cir, mincir, bc, be). Vik can you comment on that or anyone else who knows for sure. Thanks, Ryan Trauernicht
[OSL | CCIE_Voice] New School switchport config with QOS
For Campus QOS. If you configure a port on the 3550: int fa0/1 switchport mode access switchport access vlan 100 switchport voice vlan 200 spanning-tree portfast Does the mls qos trust cos still apply since it isnt a dot1q port. COS is a 802.1q header. Or would you need to change it to mls qos trust dscp? Also what is the proper configuration for an ATA. Since there is no PC port do we want to still configure it as a trunk or dedicate it to only the voice vlan? Thanks, Ryan Trauernicht
