[OSL | CCIE_Voice] Pass
Finallytook more times than I care to admit! A big thanks to IPexpert (especially Vic) and everybody who has been a part of this list. Jeff Cotter CCIE #27033 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Prompt Issue with IPCC
I have a problem with 1 prompt in a script. I created 3 prompts using Unity Connection TRAP. I played back each prompt to insure accuracy and did a Save As to desktop. Uploaded all 3 prompts to the Prompt Repository same location successfully and no errors reported. Used the Play prompt steps to call the prompts in the script. P[myrecording.wav] The issue is...one of the prompts I get nothing. Script just go right through the step...no silence no delay nothing. The other two prompts work fine. No error when running reactive debug. All prompt were recorded within the same TRAP session so all formatting should be identical. I do not have access to the actual server at this time to continue troubleshooting. The only difference I can come up with between the prompts is the prompt that was not working was larger maybe a second or two longer than the other 2 prompts that were working. (May not be important at all). I was curious if anybody could provide some possible reasons for this or direction on future troubleshooting. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] alternate methods for file upload to CUCM (Ashar Siddiqui)
I have been trying to figure this out for some time. I finally posted a question on Cisco Ask the Experts. The reply was you can't upload file to UCM via SSH only TFTP? See link below for discussion. https://supportforums.cisco.com/message/3151409#3151409 Jeff -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Monday, August 16, 2010 10:25 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 54, Issue 56 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. copy files from windows pc using windows own tftp (Stutz, Bernhard) 2. Re: copy files from windows pc using windows own tftp (Stutz, Bernhard) 3. alternate methods for file upload to CUCM (Miron Kobelski) 4. Re: alternate methods for file upload to CUCM (Ashar Siddiqui) 5. Re: Mgcp (Erwan Erwan) -- Message: 1 Date: Mon, 16 Aug 2010 18:00:50 +0200 From: Stutz, Bernhard st...@pandacom.de To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] copy files from windows pc using windows own tftp Message-ID: 8eb8e7054d698544b600adf5ef068fdb035d2...@ffmpdcexch1.pandacom.de Content-Type: text/plain; charset=iso-8859-1 Hi, where is the destination directory located when you try to download a file to a routers flash using tftp? I am not using that tiny tftp32.exe just windows own tftp service. However on that tiny tftp32.exe you can select the tftp home directory but where is the home directory on windows own tftp service? its not c:\windows\system32 where tftp.exe is located... cheers, Bernhard -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20100816/9acd48b2/attachment-0001.html -- Message: 2 Date: Mon, 16 Aug 2010 18:04:20 +0200 From: Stutz, Bernhard st...@pandacom.de To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] copy files from windows pc using windows own tftp Message-ID: 8eb8e7054d698544b600adf5ef068fdb035d2...@ffmpdcexch1.pandacom.de Content-Type: text/plain; charset=iso-8859-1 just tried the user's home directory and that's it... so: c:\document and settings\username is also the home directory of the windows own tftp service. it just came to my mind to try this when i wrote down home directory on this email... cheers, Bernhard Von: ccie_voice-boun...@onlinestudylist.com im Auftrag von Stutz, Bernhard Gesendet: Mo 16.08.2010 18:00 An: ccie_voice@onlinestudylist.com Betreff: [OSL | CCIE_Voice] copy files from windows pc using windows own tftp Hi, where is the destination directory located when you try to download a file to a routers flash using tftp? I am not using that tiny tftp32.exe just windows own tftp service. However on that tiny tftp32.exe you can select the tftp home directory but where is the home directory on windows own tftp service? its not c:\windows\system32 where tftp.exe is located... cheers, Bernhard -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20100816/0e9df452/attachment-0001.html -- Message: 3 Date: Mon, 16 Aug 2010 18:25:16 +0200 From: Miron Kobelski findko...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] alternate methods for file upload to CUCM Message-ID: aanlkti=xfpdhkq9jk714sqqfpdgkdcnn_tf72eau5...@mail.gmail.com Content-Type: text/plain; charset=utf-8 Hello, I've seen this question a few times, but I still have no clue how to do it. We can upload files to CUCM via web page, but is it possible via SSH? you can download files from CUCm via SSH, but can it be done in the other way? regards kobel -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20100816/85cd8660/attachment-0001.html -- Message: 4 Date: Mon, 16 Aug 2010 17:47:20 +0100 From: Ashar Siddiqui siddas...@gmail.com To: 'Miron Kobelski' findko...@gmail.com Cc: 'CCIE Voice OSL' ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] alternate methods for file upload to CUCM Message-ID: 00b301cb3d62$b4d26dd0$1e7749...@com Content-Type: text/plain; charset=utf-8 Count me in too. I am also
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 54, Issue 3
I am thinking using the bandwidth limit command would limit ALL queues to that rate not just the priority queue. Also see below response from ask the experts stating buffers and thresholds have no effect on bandwidth. I am thinking now maybe to adjust the INGRESS priority queue bandwidth setting. Use the srr-queue shape command on the trunk connecting to the router and use the priority queue out command on the ports connecting the phones. The answer is NO. you cannot have priority queue enabled and give the Q only 25% bandwidth. When Priority Q is enabled the Q gets 100% bandwidth. If there are any packets in the Q all those packets will be serviced before getting to the other queues. you can use mls qos queue-set output qset-id threshold queue-id drop-threshold1 drop-threshold2 reserved-threshold maximum-threshold command only set the buffer size and/or threshold and it has not nothing to do with bandwidth. -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Monday, August 02, 2010 9:00 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 54, Issue 3 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: Layer 2 QOS (Beck, Ken) 2. Re: ip phone on layer 3 interface (ShinGei Yong) 3. mls qos queue-set output qset-id thresholdqueue-id drop-threshold1 drop-threshold2 , what are these two drop thresholds? (jeremy co) -- Message: 1 Date: Sun, 1 Aug 2010 14:09:44 -0700 From: Beck, Ken kb...@vectorusa.com To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Layer 2 QOS Message-ID: a18cfc04cd8374489edf3ccc53f72c49830...@vlamail02.vector.djjr Content-Type: text/plain; charset=us-ascii What about if we just set the egress limit to 10% of the total bandwidth of the port like this below. Limiting the Bandwidth on an Egress Interface Switch(config)# interface FastEthernet1/0/2 Switch(config-if)# srr-queue bandwidth limit 10 When you configure this command to 80 percent, the port is idle 90 percent of the time. The line rate drops to 10 percent of the connected speed, which is 10 Mb/s. These values are not exact because the hardware adjusts the line rate in increments of six. Does that work for the question? Regards, Ken Beck -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Thursday, July 29, 2010 7:06 PM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 53, Issue 142 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. Layer 2 QOS (Jeff Cotter) 2. Re: Layer 2 QOS (Matthew Berry) 3. Re: Layer 2 QOS (Jeff Cotter) 4. Re: Layer 2 QOS (Daniel Berlinski) 5. Re: Layer 2 QOS (Matthew Berry) -- Message: 1 Date: Thu, 29 Jul 2010 18:19:19 -0700 From: Jeff Cotter jcot...@voxns.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Layer 2 QOS Message-ID: 54cc1bd3093b6e41b86926c1657432f1a6264...@ssfex1 Content-Type: text/plain; charset=us-ascii How would you enable the priority queue AND make sure queue 1 has 10% of the bandwidth. The documentation states that if the priority queue in enabled, shape and share configuration for that queue is ignored. So how do you accomplish this without using Shape command. -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20100729/fb486bb2/attachment-0001.html -- Message: 2 Date: Thu, 29 Jul 2010 20:47:35 -0500 From: Matthew Berry ciscovoiceg...@gmail.com To: Jeff Cotter jcot...@voxns.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL
[OSL | CCIE_Voice] SNMP Master Agent on VMware
Anybody know if there is a work around for this. Trying to test some 3rd party apps on my Call Manager which is running on VMware Server and they need the SNMP Master Agent started. You'll notice that SNMP Master Agent service fails to start if CUCM was installed on VMWare ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Layer 2 QOS
How would you enable the priority queue AND make sure queue 1 has 10% of the bandwidth. The documentation states that if the priority queue in enabled, shape and share configuration for that queue is ignored. So how do you accomplish this without using Shape command. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Layer 2 QOS
Hello Matthew and thanks for the reply. However my thought is……putting COS 5 and EF into Q1 does not make it a priority queue. Which by definition means the queue is serviced until it is empty BEFORE the other queues are serviced. This behavior is only in effect with the priority queue out command. The Ingress queues have the proper commands to control the size of the priority queue mls qos srr-queue input priority-queue queue-id bandwidth weight but not the egress queues. Of course I my logic could be flawed here. From: Matthew Berry [mailto:ciscovoiceg...@gmail.com] Sent: Thursday, July 29, 2010 6:48 PM To: Jeff Cotter Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Layer 2 QOS You could treat it as a priority queue by throwing CoS 5 or DSCP EF into Q1. You could then shape it to 10, which would result in 10%. You would also need to do a no priority-queue out under the interface. But I don't think you can have priority on the queue and still limit the queue to only part of the pipe. Matthew Berry **Sent from my iPhone** Skype/Twitter: ciscovoiceguru Google Voice: +1 612 424 5044 On Jul 29, 2010, at 20:19, Jeff Cotter jcot...@voxns.commailto:jcot...@voxns.com wrote: How would you enable the priority queue AND make sure queue 1 has 10% of the bandwidth. The documentation states that if the priority queue in enabled, shape and share configuration for that queue is ignored. So how do you accomplish this without using Shape command. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Layer 2 QOS
Interesting..Thanks Daniel great thought! From: Daniel Berlinski [mailto:dberlin...@gmail.com] Sent: Thursday, July 29, 2010 7:03 PM To: Jeff Cotter Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Layer 2 QOS In my opinion this is done by adjusting the buffer size for queue 1 and applying it to a queue-set. srr shape statement in my opinion means nothing in relation to adjusting priority queue size. http://onlinestudylist.com/archives/ccie_voice/2010-July/069398.html On Fri, Jul 30, 2010 at 1:19 PM, Jeff Cotter jcot...@voxns.commailto:jcot...@voxns.com wrote: How would you enable the priority queue AND make sure queue 1 has 10% of the bandwidth. The documentation states that if the priority queue in enabled, shape and share configuration for that queue is ignored. So how do you accomplish this without using Shape command. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Number of CME Conference DNs needed.
Looking for confirmation on calculating the number of conference directory numbers needed on cme, for either addhoc or meetme as I do not have enough phones to test myself. My logic would be to take the Maximum Sessions X the Maximum Conference-Parties configured under the dspfarm profile for conferencing. For instance, example below would require 12channels either 6 dual-lines in hunt... or 2 octolines in hunt. Please let me know if you agree or disagree with this logic. Dspfarm profile 1 conference Codec g711u Codec g729r8 Maximum sessions 3 Maximum conference-parties 4 Associate application sccp Admin guide also indicates the maximum conference-parties command is specific to meet-me with no mention of addhoc. Is this a valid statement? Thanks calculate the number of conference DNs required based on the Maximum-Conference-Participants configured in the DSP Profile as well as the Maximum Sessions. I am assuming each phone that initiates a conf either add hoc or Meetme is considered a Session ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CUE MWI
I noticed the ipexpert material use out calling for mwi when CUE integrates with UCME. Was curious why, as the documentation recommends sub-notify or unsolicited. Is there some gotcha we should know about? See below from admin guide. The outcall option is available for backward compatibility. We recommend that you use either sub-notify or unsolicited for the MWI notification option. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Direct transfer to Original called party Voicemail
I believe someone already suggested this but why not just use the transfer to voicemail key. See below from Admin Guide. The Transfer to Voice Mail feature allows a phone user to transfer a caller directly to a voice-mail extension. The user presses the TrnsfVM soft key to place the call on hold, enters the extension number,and then commits the transfer by pressing the TrnsfVM soft key again. The caller hears the complete voice mail greeting. This feature is supported using the TrnsfVM soft key or feature access code (FAC). -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Wednesday, May 26, 2010 5:39 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 51, Issue 157 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: Direct transfer to Original called party Voicemail (Angel Perez) 2. Re: Direct transfer to Original called party Voicemail (Rogers Ochieng) -- Message: 1 Date: Wed, 26 May 2010 12:27:31 + From: Angel Perez gorr...@hotmail.com Subject: Re: [OSL | CCIE_Voice] Direct transfer to Original called party Voicemail To: siddas...@gmail.com, r.ochi...@mfient.com Cc: osl osl ccie_voice@onlinestudylist.com Message-ID: col110-w59e42d65e1e2d07e3cddd9a1...@phx.gbl Content-Type: text/plain; charset=windows-1252 For all these extension wouldn't be scalable... I think that this behaviour could be changed system wyde but I can't remember how From: siddas...@gmail.com To: r.ochi...@mfient.com Date: Wed, 26 May 2010 12:03:46 +0100 CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Direct transfer to Original called party Voicemail So you mean in CUE, I just have to assign alternate extension for every user starting with 6 like 62904, 62905, 62906 ... If x2905 transfer the call to 62904, would it go straight to VM for 2904 or will it first ring for 10s and then go to voicemail? Do I have to create ephone-dn for all of these? (remember customer has 100+ users and dn) Thanks for your help Ash From: Rogers Ochieng [mailto:r.ochi...@mfient.com] Sent: 26 May 2010 10:57 To: 'Ashar Siddiqui' Subject: RE: [OSL | CCIE_Voice] Direct transfer to Original called party Voicemail I?m thinking secondary number in CUE for the user say 62904 and you route that to CUE so 6 can be your assumed prefix for diverting calls to CUE for other subscriber From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ashar Siddiqui Sent: Wednesday, May 26, 2010 12:20 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Direct transfer to Original called party Voicemail Hello all, One of my customer is interested in direct transfer of an incoming call to voicemail after it has been picked up by someone else in the pickup group. For e.g. If a call comes in and ring x2904 but he is not available, person at x2905 picks up the call but the calling party wants to leave a VM for x2904. How the person at x2905 can direct transfer the call to x2904 voicemail. One way is to transfer the call back to x2904 which will ring and ring for 10s and then go to voicemail. This is not what they want. They want the ability to transfer the call directly to voicemail of Original called party. ephone-dn 1 octo-line number 2904 pickup-group 1 label Tim Flynn (2904) name Tim Flynn call-forward busy 8005 call-forward noan 8005 timeout 10 corlist incoming User-international ! ! ephone-dn 2 octo-line number 2905 pickup-group 1 label Steve Zander (2905) name Steve Zander call-forward busy 8005 call-forward noan 8005 timeout 10 corlist incoming User-international ! ! Ash _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. https://signup.live.com/signup.aspx?id=60969 -- next part -- An HTML attachment was scrubbed... URL: http://onlinestudylist.com/pipermail/ccie_voice/attachments/20100526/40951836/attachment-0001.htm -- Message: 2 Date: Wed, 26 May 2010 15:37:19 +0300 From: Rogers Ochieng r.ochi...@mfient.com Subject: Re: [OSL | CCIE_Voice] Direct transfer to Original called
[OSL | CCIE_Voice] Expensive Lunch
Sat the lab yesterdayfailed. So I need to vent a little bit. There is no doubt I failed the exam and it was painfully obvious I am not ready to be a CCIE. However the most unsettling piece is not getting the points in areas that I thought were working and verified. I do not know how to address this the next time. My fear is, I will probably program these items exactly the same way next time... because that is the way I know howthey coincide with the training materials available and most importantly they seem to work and I will not get the points AGAIN!! I just do not understand why I did not get these points or how to fix it the next timefrustrated. Lunch was good though! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Unable to Bind L3 to CCM (Jeff Price (jeffpric))
You need to add the Service-MGCP after your PRI group. Then you should be able to bind l3 -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Friday, May 21, 2010 3:55 PM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 51, Issue 120 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: translation rule (David Holman) 2. Re: translation rule (Ashar Siddiqui) 3. Unable to Bind L3 to CCM (Jeff Price (jeffpric)) -- Message: 1 Date: Fri, 21 May 2010 16:18:22 -0400 From: David Holman davidkhol...@gmail.com Subject: Re: [OSL | CCIE_Voice] translation rule To: Wael Agina waelag...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Message-ID: aanlktimdrfpmqhxsh-debrbuatcymvo0gknjiosnr...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 I keep this link handy for voice translation questions: http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0080325e8e.shtml On Fri, May 21, 2010 at 4:15 PM, Wael Agina waelag...@gmail.com wrote: Dear Ashar, The ^$ is catching null, which could be used to catch calls from unkown. example usage, drop any calls from PSTN that has ANI of unkown type. On H323 you could use following rule to do this voice translation-rule 1 rule 1 reject /^$/ voice translation-profile Drop-Unknown translate calling 1 dial-peer voice 1 pots direct-inward-dial incom called . *call-block translation-profile incoming Drop-Unknown* For you example may be it i setting unknown ANI to be 42000 for example, bu not sure, need to be tested. Regards, Wael Agina On Fri, May 21, 2010 at 11:02 PM, Ashar Siddiqui siddas...@gmail.comwrote: Hi, I know I may sound stupid to some but I really want to know the purpose of ^$ in a translation rule for e.g: voice translation-rule 100 rule 1 /^$/ /42000/ ! ^$ is null...what does it mean? what is a null number? Ash ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Thanks and Best Regards, Wael Agina ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- next part -- An HTML attachment was scrubbed... URL: http://onlinestudylist.com/pipermail/ccie_voice/attachments/20100521/3259ccc1/attachment-0001.htm -- Message: 2 Date: Fri, 21 May 2010 21:58:27 +0100 From: Ashar Siddiqui siddas...@gmail.com Subject: Re: [OSL | CCIE_Voice] translation rule To: David Holman davidkhol...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Message-ID: 4bf6f3f3.4060...@gmail.com Content-Type: text/plain; charset=us-ascii An HTML attachment was scrubbed... URL: http://onlinestudylist.com/pipermail/ccie_voice/attachments/20100521/9a547d02/attachment-0001.htm -- Message: 3 Date: Fri, 21 May 2010 17:55:16 -0500 From: Jeff Price (jeffpric) jeffp...@cisco.com Subject: [OSL | CCIE_Voice] Unable to Bind L3 to CCM To: CCIE Voice Maillist ccie_voice@onlinestudylist.com Message-ID: b2de0afa86565c47bd3a8435550f955301068...@xmb-rcd-201.cisco.com Content-Type: text/plain; charset=us-ascii Hey everyone, Have you ever seen a situation where you can register a MGCP GW to CUCM but you are unable to bind L3 to CCM in IOS? Here's what I see: R1(config-if)#isdn bind-l3 ? q931 Select IOS Q.931 R1(config-if)# Here is my config: R1#show run Building configuration... Current configuration : 3127 bytes ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec service password-encryption ! hostname R1 ! boot-start-marker boot system flash:c2800nm-adventerprisek9_ivs_li-mz.124-24.T.bin boot-end-marker ! logging message-counter syslog enable password 7 110A1A0C12 ! no aaa new-model network-clock-participate wic 0 ! dot11 syslog ip source-route ! ! ip cef ip dhcp excluded-address 10.5.200.1 ! ip dhcp pool HQ_PHONES network 10.5.200.0 255.255.255.0 option 150 ip 172.21.51.204 default-router 10.5.200.1 ! ! no ipv6 cef ! multilink bundle-name authenticated ! ! !
[OSL | CCIE_Voice] Another Cube issue
I am going crazy. Call from Call Manager to CME via GK/cube. SCCP phones only. G729 from end to end. CME phone rings but when I answer call drops. I have unchecked Wait for h245. I have selected outbound fast start and registered and MTP with g729r8 configured. (I do not believe MTP should be required) I have a transcoder registered with both CME and CUBE but again call is g729 end to end. Relevant config from CME below. dspfarm profile 10 mtp codec pass-through codec g729r8 maximum sessions software 10 associate application SCCP dial-peer voice 333 voipinbound voip dial-peer for cube call to ext 3001 incoming called-number 3001 dtmf-relay h245-alphanumeric no vad R2801#sh gatek calls Total number of active calls = 2. GATEKEEPER CALL INFO LocalCallIDAge(secs) BW 31-168 2 16(Kbps) Endpt(s): Alias E.164Addr src EP: gk-trunk_11006 CallSignalAddr Port RASSignalAddr Port 10.1.100.21 40649 10.1.100.21 32808 Endpt(s): Alias E.164Addr dst EP: cube 3001 CallSignalAddr Port RASSignalAddr Port 192.168.1.9 1720 192.168.1.9 54941 LocalCallIDAge(secs) BW 32-168 2 16(Kbps) Endpt(s): Alias E.164Addr src EP: cube 1006 CallSignalAddr Port RASSignalAddr Port 192.168.1.9 1720 192.168.1.9 54941 Endpt(s): Alias E.164Addr dst EP: cme 3001 CallSignalAddr Port RASSignalAddr Port 3.3.3.3 1720 3.3.3.3 65239 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Transcoder and BACD issue
Station to Station calls work as expected between UCM and CME including g729 call through gatekeeper to CME sip phone using g711. Transcoder is invoked properly and confirmed with show sccp connections. Call Routing and Transcoder proven as functional with above. However call fails immediately if I put my GK Trunk on UCM to g729 region (due to codec mismatch). If I change CME BACD dial-peer to g729 you hear silence. Voice-class codec on BACD dial-p you hear silence. Change UCM Trunk to g711u region and BACD dial peer to g711u call works fine. How is one supposed to get calls over the WAN using g729 to work with BACD?? Dial-peer config below. All other config is proven to be working dial-peer voice 222 voip service aa destination-pattern 3500 session target ipv4:3.3.3.3 (loopback interface) incoming called-number 3500 dtmf-relay h245-alphanumeric codec g711ulaw no vad ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Transcoder and BACD issue
Already done and calls to and from sip work fine. The problem as I see it, is the dial-peer 222 acts as both incoming and outgoing dial-peer and IS hardcoded for g711u. Therefore a transcoder will never be invoked#!! As both inbound and outbound dial-peer is g711u so no codec mismatch no transcoder. The only solution I found is to call a different number say 3800 from UCM call routes to CME hits dial-peer 3800 dial-peer with g729 codec configured!! I than set up ephone-dn 3800 and call-forward all to BACD pilot number of 3500. This works. Again I see no possible way of transcoding this BACD dial-peer configured as documentation states because it acts as both legs of the call and hardcoded to g711u so no way is a transcoder going to be invoked. From: vccie2010 [mailto:vccie2...@gmail.com] Sent: Friday, April 30, 2010 12:50 PM To: Jeff Cotter Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Transcoder and BACD issue ON your GK trunk - Uncheck H245 box On Fri, Apr 30, 2010 at 12:20 PM, Jeff Cotter jcot...@voxns.commailto:jcot...@voxns.com wrote: Station to Station calls work as expected between UCM and CME including g729 call through gatekeeper to CME sip phone using g711. Transcoder is invoked properly and confirmed with show sccp connections. Call Routing and Transcoder proven as functional with above. However call fails immediately if I put my GK Trunk on UCM to g729 region (due to codec mismatch). If I change CME BACD dial-peer to g729 you hear silence. Voice-class codec on BACD dial-p you hear silence. Change UCM Trunk to g711u region and BACD dial peer to g711u call works fine. How is one supposed to get calls over the WAN using g729 to work with BACD?? Dial-peer config below. All other config is proven to be working dial-peer voice 222 voip service aa destination-pattern 3500 session target ipv4:3.3.3.3 (loopback interface) incoming called-number 3500 dtmf-relay h245-alphanumeric codec g711ulaw no vad ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] UCM Personal Directory Error
I get an error contact system administrator when I select Directories button than Personal Directory or Corporate Directory for that matter. (no ldap synchronization in place. Users have been added locally in UCM) I have the URL from PUB to IP address for all services under Enterprise Parameters, the phone service for personal directory is the default setting and is enabled. Phones are 7965s I can access the user web page and edit the PAB but can't access from Phone. Looking for some direction on troubleshooting this. Thanks Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCM Personal Directory Error
Thanks, I did read that but does not help. From: vccie2010 [mailto:vccie2...@gmail.com] Sent: Thursday, April 29, 2010 10:36 AM To: Jeff Cotter Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] UCM Personal Directory Error Try this out...might be helpful http://docs.google.com/viewer?a=vq=cache:tOqpVoOf8BUJ:www.ciscocertified.info/application/pdf/paws/109369/fix_issucorp.pdf+error+contact+system+administrator+cisco+cucm+personal+directoryhl=engl=uspid=blsrcid=ADGEEShqjNZGHmBt9hSwKO_HCeLklQZy6pPZALq1fZkTSsZNtw5H72r6eNU9lGBvuC-Y75srZ_TM9suBH4T1R7XFX9iWn6GDNZZn6eMIVdIFnIY7VAb3BOuIpSRNYF4YeAxx774MyES9sig=AHIEtbREtLDh9h3R-koPVhTUQxA0py0axg On Thu, Apr 29, 2010 at 10:05 AM, Jeff Cotter jcot...@voxns.commailto:jcot...@voxns.com wrote: I get an error contact system administrator when I select Directories button than Personal Directory or Corporate Directory for that matter. (no ldap synchronization in place. Users have been added locally in UCM) I have the URL from PUB to IP address for all services under Enterprise Parameters, the phone service for personal directory is the default setting and is enabled. Phones are 7965s I can access the user web page and edit the PAB but can't access from Phone. Looking for some direction on troubleshooting this. Thanks Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] FRF12 Calculation Clarification
There is a huge number of posts on this, and explanations are all over the board. Trying to get a definitive answer. When calculating the size of LLQ AND utilizing FRF.12. Is the Layer 2 overhead 4 bytes for Frame Relay or 8 Bytes. SRND states 8 Solution Guides appear to use 4 or 7. One post indicates if configured properly voice packets will not be fragmented therefore use standard Frame Relay overhead of 4 bytes. This seems logicalThanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol2 Lab1 4.2 issue (ccieid1ot)
Not saying this is your issue but you should remove the Voice-Class codec from the dial-peers involved in GK and Cube call, hardcode the codec in ALL the dial-peers and see if this helps you. -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Thursday, April 22, 2010 8:25 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 50, Issue 123 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: Vol2 Lab1 4.2 issue (ccieid1ot) -- Message: 1 Date: Thu, 22 Apr 2010 10:25:00 -0500 From: ccieid1ot ccieid...@gmail.com Subject: Re: [OSL | CCIE_Voice] Vol2 Lab1 4.2 issue To: Angel Perez gorr...@hotmail.com Cc: osl osl ccie_voice@onlinestudylist.com, kevin.hobson2...@ntlworld.com, ccie_voice-requ...@onlinestudylist.com Message-ID: w2te51ace101004220825la0c4627bof96d74e8f9a18...@mail.gmail.com Content-Type: text/plain; charset=ISO-8859-1 Did you add G729r8 in DSPfarm profile? On Thu, Apr 22, 2010 at 9:34 AM, Angel Perez gorr...@hotmail.com wrote: Hi,?I'm not 100% sure?of your?scenario but try the?following tips: On gk trunk: uncheck?wait for h245?capabilies, chech inbound faststart On cube: voice service voip h323 emptycapability h225 id-passthru h225 connect-passthru h245 passthru tcsnonstd-passthru add codec g729r8 to dspfarm profile On remote gw: If your phones are sip? with codec g711u you will need a transcoder local to this gw hth Date: Thu, 22 Apr 2010 15:16:41 +0100 From: kevin.hobson2...@ntlworld.com To: ccie_voice@onlinestudylist.com CC: ccie_voice-requ...@onlinestudylist.com Subject: [OSL | CCIE_Voice] Vol2 Lab1 4.2 issue Hi all, I have been banging my head against this for a few hours now. I have a issue were if you call a br2 phone from hq the call routes fine and the call gets setup. However when the call is answered the call continues to ring on the hq side and the call is disconnected on the outbound call leg with cause 47 no resource available. I have narrowed this down to a transcoding issue as if i change outbound dialpeer to g729 the call connects fine. If i do sh sdspfarm session active i dont see anything so it looks like for some reason the CUBE isnt invoking the transcoder. A show sdspfarm unit show that it is registered: GK-CUBE#sh sdspfarm units mtp-1 Device:xcoder TCP socket:[1] REGISTERED in SCCP ver 0/10 actual_stream:4 max_stream 12 IP:10.10.200.2 48984 MTP YOKO keepalive 17 Supported codec: G711Ulaw G711Alaw G729 G729a G729ab max-mtps:1, max-streams:4, alloc-streams:4, act-streams:0 Debug voip ipipgw below: GK-CUBE# Apr 22 18:08:35.095: //9/8041DD601500/H323/setup_ind: Receive bearer cap infoXRate 16, rateMult 0 Apr 22 18:08:35.103: //9/8041DD601500/H323/cch323_set_h245_state_mc_mode_incoming: h245 state m/c mode=0x10F, h323_ctl=0x2F Apr 22 18:08:35.115: //-1//H323/cch245_event_handler: callID=9 Apr 22 18:08:35.115: //-1//H323/cch245_event_handler: Event CC_EV_H245_SET_MODE: data ptr=0x4AC70D60 Apr 22 18:08:35.115: //9/8041DD601500/H323/cch323_set_mode: callID=9, flow Mode=1 spi_mode=0x1 Apr 22 18:08:35.115: //9/8041DD601500/H323/cch323_do_call_proceeding: set_mode NOT called yet...saved deferred CALL_PROC Apr 22 18:08:35.115: //10/8041DD601500/H323/cch323_set_h245_state_mc_mode_outgoing: call_spi_mode = 1 Apr 22 18:08:35.115: //10/8041DD601500/H323/cch323_set_h245_state_mc_mode_outgoing: h245 state m/c mode=0x1AF0, h323_ctl=0x0 Apr 22 18:08:35.115: //10/8041DD601500/H323/cch323_get_peer_info: Entry Apr 22 18:08:35.115: //10/8041DD601500/H323/cch323_get_peer_info: Have peer Apr 22 18:08:35.115: //10/8041DD601500/H323/cch323_set_pref_codec_list: First preferred codec(bytes)=5(160) Apr 22 18:08:35.115: //10/8041DD601500/H323/cch323_get_peer_info: Flow Mode set to FLOW_THROUGH Apr 22 18:08:35.115: //10/8041DD601500/H323/cch323_set_h323_control_options_outgoing: h245 sm mode = 6896 Apr 22 18:08:35.115: //10/8041DD601500/H323/cch323_set_h323_control_options_outgoing: h323_ctl=0x2F Apr 22 18:08:35.119: //9/8041DD601500/H323/cch323_process_set_mode: Setting inbound leg mode flags to 0x1AF0, flow-mode to FLOW_THROUGH Apr 22 18:08:35.119: //9/8041DD601500/H323/cch323_process_set_mode: Sending deferred CALL_PROC Apr 22 18:08:35.119:
Re: [OSL | CCIE_Voice] Converting CUE to Integrate with UCM
Yes, this is the only requirement. I have performed it a couple of times now with no issues moving back and forth between the two types of integration. From: Berry, Matthew J. [mailto:mjbe...@krollontrack.com] Sent: Wednesday, April 21, 2010 11:04 AM To: Jeff Cotter; osl osl Subject: RE: Converting CUE to Integrate with UCM Jeff - Did you ever find out the answer to your question? I'm curious. Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.commailto:david.ra...@kroll.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Cotter Sent: Monday, March 22, 2010 6:12 PM To: osl osl Subject: [OSL | CCIE_Voice] Converting CUE to Integrate with UCM Is the only requirement to go from CME integration to UCM to load the proper license file? This is my companies equipment not proctor labs. I would like to be able to move back and forth similar to proctor labs but am unsure it is as easy as just loading the proper license file. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CUE integration with multiple CMEs
Looking for some clarification on support for multiple CME sites with a single CUE module and provide MWI notification to remote sites. Release notes for 3.1 indicate support for integration to multiple CME however admin guide and design guides state the following: Restrictions for Integrating Cisco CME with Cisco Unity Express Cisco Unity Express cannot provide voice-mail services across Cisco CME routers. Cisco Unity Express can provide voice mail services only for phones on its host Cisco CME router. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Transcoder AGAIN!
I can't seem to get the voice-class command and transcoder to work between UCM and CME SIP phone. If I explicitly Configure the codec on the dial-peer to match the UCM Trunk region setting call completes and xcoder on CME is invoked properly. Remove the codec command and put voice-class command in and call fails every time. This holds true if call is SIP, H323 trunk, Gatekeeper Trunk with or without CUBE. I do NOT have a xcoder configured on my UCM for this scenario (due to hardware limitation on my home lab). I do not believe this is required as phones natively support both g729 and g711 however please correct me if I am mistaken. Call between UCM and CME via an h225 GK trunk. No CAC configured on GK. GK trunk configured in a g729 only region on UCM. Incoming Dial-Peer on CME configured with Voice-Class Codec 1. Voice-Class contains both g711u and g729r8. CME Phone is running SIP, G711 Codec selected under Voice Register Pool. Transcoder configured on CME and registered with telephony-service. If I originate the call from the SIP phone to UCM via GK call completes. If I originate the call from the UCM phone call fails after answer. (transcoder not being invoked) If I REMOVE the Voice-Class from the incoming dial-peer on CME and replace with codec g729r calls complete and transcoder in invoked properly. I do not understand why the Voice-Class command is affecting the Transcoder operation? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] g729 MOH
I was just testing in my lab. I also see that the router is sourcing music on 239.1.1.3 when I do a show IP mroute, however I just hear silence. I also believe Mark Snow indicates on VOD you can tftp g729 music file from UCM to router flash and use that.Not sure how to locate the file and name on UCM still working on that. Thanks for the responses. Jeff From: Angel Perez [mailto:gorr...@hotmail.com] Sent: Wednesday, March 17, 2010 6:55 AM To: earl.ho...@pcmall.com; arunv...@gmail.com; Jeff Cotter Cc: osl osl Subject: RE: [OSL | CCIE_Voice] g729 MOH Hi Earl: I suppose that the music-on-hold.au file loaded at flash should be g729 formatted for this case? (and also gw and ccm are configured correctly for g729 codec) But my question is: If you are playing moh from router and it is playing only localy, why should you want to play moh as g729 instead of g711? Or it was just a test you done? Thanks Date: Wed, 17 Mar 2010 09:41:12 -0400 From: earl.ho...@pcmall.com To: arunv...@gmail.com; jcot...@voxns.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] g729 MOH I agree that Cisco's documentation states that when spoofing from flash, only G711 is supported. However, I have proven this not to be true in production. The sound quality isn't great, but when doing show ccm-manager music-on-hold it is obvious that the source MAC address is 239.1.1.3, which would be appropriate for G729. Earl Hough CCIE #16508 (RS/Security) From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Arun Kumar Sent: Wednesday, March 17, 2010 12:22 AM To: Jeff Cotter Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] g729 MOH Router only supports G711 format moh not g729. On Wed, Mar 17, 2010 at 2:33 AM, Jeff Cotter jcot...@voxns.commailto:jcot...@voxns.com wrote: I have both g729 and g711 Multicast MOH working using the UCM MOH server so I know my layer 3 interfaces are passing multicast traffic. My problem is when I use one of my branch routers as the source for the music (set max hops on UCM MOH server to 1 and source multicast from router using Telephony Service Mulitcast MOH 239.1.1.3 port 16384 etc.) I can only get g711 to work. I suspect that the music file on my router flash only supports g711.am I on the right track here if so is there a work around?? Thank you. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ The information contained in this transmission is confidential. It is intended solely for the use of the individual(s) or organization(s) to whom it is addressed. Any disclosure, copying or further distribution is not permitted unless such privilege is explicitly granted in writing by PC Mall, Inc. Furthermore, PC Mall, Inc. is not responsible for the proper and complete transmission of the substance of this communication, nor for any delay in its receipt. Hotmail: Trusted email with powerful SPAM protection. Sign up now.https://signup.live.com/signup.aspx?id=60969 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] g729 MOH
I have both g729 and g711 Multicast MOH working using the UCM MOH server so I know my layer 3 interfaces are passing multicast traffic. My problem is when I use one of my branch routers as the source for the music (set max hops on UCM MOH server to 1 and source multicast from router using Telephony Service Mulitcast MOH 239.1.1.3 port 16384 etc.) I can only get g711 to work. I suspect that the music file on my router flash only supports g711.am I on the right track here if so is there a work around?? Thank you. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] via gatekeeper invia key word
Thanks Otto, if this is the case then I believe the explanation Mark S. gives on the VOD is incorrect. As he references the invia between local zones. From: Otto Sanchez [mailto:o...@ipexpert.com] Sent: Friday, March 12, 2010 6:14 AM To: Jeff Cotter Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] via gatekeeper invia key word Hi Jeff, According to your lab results, you are describing the expected behavior, more information at: http://www.cisco.com/en/US/partner/docs/ios/voice/cubegk/configuration/guide/ve-gk-config_ps6441_TSD_Products_Configuration_Guide_Chapter.html#wp1225776 Thanks!, On Thu, Mar 11, 2010 at 9:55 AM, Jeff Cotter jcot...@voxns.commailto:jcot...@voxns.com wrote: I am struggling a bit with the invia concept. I think I understand the outvia. When I lab this up I find the following. Invia only applies to calls coming from a remote GK. In order for call to use cube I had to configure the invia key word on the actual remote zone.not on the destination zone. Sample config of my invia GK gk zone local ucm cisco.comhttp://cisco.com 1.1.1.1 gk zone local cube gk zone local cme gk zone remote gk2 lab.comhttp://lab.com 2.2.2.2 invia cube zone prefixs omitted So calls coming FROM gk2 destined for either ucm or cme zone used the cube. If I applied the invia key word on either ucm or cme zone directly, the cube was not invoked. This seems to conflict with the proctor guide mock lab 1 statement invia command when defining the UCME zone would invoke the cube for calls coming in from a remote zone. In my lab applying invia directly to destination zone had no affect and cube was not invoked. Am I missing something. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP Hardware Transcoder
FYI, I was only able to get this to work using transcoder on CME. Had to match the codec between UCM trunk and incoming dial-peer on CME...then xcoder would engage on CME for the SIP phone. I have a hardware limitation in my home lab so I am not able to configure a xcoder on both UCM and CME simultaneously. From: Omotayo [mailto:adefilabi...@gmail.com] Sent: Friday, March 12, 2010 6:33 AM To: Otto Sanchez Cc: Jeff Cotter; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SIP Hardware Transcoder Hello Otto, i had same issue The transcoder can be on the trunk? When i did the transcoder on the br2 router, i get a busy tone when the sip phone is being called from the hq phone REgards On Fri, Mar 12, 2010 at 12:59 PM, Otto Sanchez o...@ipexpert.commailto:o...@ipexpert.com wrote: Hi Jeff, Would you please tell us more about the call flow and the end to end codec requirements for this call. If doing g.729 over the wan, and your sip phone is using g.711 you should transcode at br2, Please let us know, Thanks, On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter jcot...@voxns.commailto:jcot...@voxns.com wrote: Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM. Can't seem to get a call from Call Manager to CME sip phone working. I can call from CME to UCM but not the other way around. Rings but disconnects when answered. Transcoder shows registered in Call manager. Thanks Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.comhttp://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] via gatekeeper invia key word
I am struggling a bit with the invia concept. I think I understand the outvia. When I lab this up I find the following. Invia only applies to calls coming from a remote GK. In order for call to use cube I had to configure the invia key word on the actual remote zone.not on the destination zone. Sample config of my invia GK gk zone local ucm cisco.com 1.1.1.1 gk zone local cube gk zone local cme gk zone remote gk2 lab.com 2.2.2.2 invia cube zone prefixs omitted So calls coming FROM gk2 destined for either ucm or cme zone used the cube. If I applied the invia key word on either ucm or cme zone directly, the cube was not invoked. This seems to conflict with the proctor guide mock lab 1 statement invia command when defining the UCME zone would invoke the cube for calls coming in from a remote zone. In my lab applying invia directly to destination zone had no affect and cube was not invoked. Am I missing something. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] SIP Hardware Transcoder
Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM. Can't seem to get a call from Call Manager to CME sip phone working. I can call from CME to UCM but not the other way around. Rings but disconnects when answered. Transcoder shows registered in Call manager. Thanks Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CME Presence
I can get this working between SCCP phones as well as having a SIP phone watch a SCCP phone. However when I configure a SCCP phone to watch a SIP phone does not work. Config below Voice Register Pool 1 works great. Ephone 1 does not work when watching Voice Register Pool 1 but does work when configured to watch ephone 2. voice register dn 1 number 3005 call-forward b2bua busy 3099 call-forward b2bua mailbox 3005 call-forward b2bua noan 3099 timeout 20 allow watch name sip phone3 mwi ! voice register pool 1 id mac 0023.EB53.27D4 type 7965 number 1 dn 1 presence call-list dtmf-relay rtp-nte sip-notify username user3 password cisco codec g711ulaw blf-speed-dial 1 3001 label sccp presence presence call-list max-subscription 128 watcher all allow subscribe sip-ua retry invite 2 timers trying 200 mwi-server ipv4:192.168.1.8 expires 120 port 5060 transport udp presence enable ephone-dn 1 dual-line number 3001 no-reg primary allow watch call-forward night-service 3099 night-service bell ephone 1 no multicast-moh mac-address 0023.EB53.26CD username user1 password cisco presence call-list fastdial 1 6509891234 name home blf-speed-dial 1 3005 label sip speed-dial 1 914089891234 label home paging-dn 20 type 7965 button 1:2 2:18 3:5 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME Presence Solved!
Actually had to make a call from the SIP phone. As soon as you break dial-tone presence indicator came on. I was just lifting the handset prior to this...another quirk with CME and SIP. From: Jeff Cotter Sent: Tuesday, March 09, 2010 3:19 PM To: 'ccie_voice@onlinestudylist.com' Subject: CME Presence I can get this working between SCCP phones as well as having a SIP phone watch a SCCP phone. However when I configure a SCCP phone to watch a SIP phone does not work. Config below Voice Register Pool 1 works great. Ephone 1 does not work when watching Voice Register Pool 1 but does work when configured to watch ephone 2. voice register dn 1 number 3005 call-forward b2bua busy 3099 call-forward b2bua mailbox 3005 call-forward b2bua noan 3099 timeout 20 allow watch name sip phone3 mwi ! voice register pool 1 id mac 0023.EB53.27D4 type 7965 number 1 dn 1 presence call-list dtmf-relay rtp-nte sip-notify username user3 password cisco codec g711ulaw blf-speed-dial 1 3001 label sccp presence presence call-list max-subscription 128 watcher all allow subscribe sip-ua retry invite 2 timers trying 200 mwi-server ipv4:192.168.1.8 expires 120 port 5060 transport udp presence enable ephone-dn 1 dual-line number 3001 no-reg primary allow watch call-forward night-service 3099 night-service bell ephone 1 no multicast-moh mac-address 0023.EB53.26CD username user1 password cisco presence call-list fastdial 1 6509891234 name home blf-speed-dial 1 3005 label sip speed-dial 1 914089891234 label home paging-dn 20 type 7965 button 1:2 2:18 3:5 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Device Mobility with Local Route Groups
Thanks Otto. From: Otto Sanchez [mailto:o...@ipexpert.com] Sent: Monday, February 22, 2010 4:46 AM To: Jeff Cotter Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Device Mobility with Local Route Groups Hi Jeff, You are right, phones at the roaming location will get the LRG configuration from the roaming DP, we cannot avoid this. However, a workaround to your scenario may be to implement + dialing, in which you globalize the caller input (translation patterns) depending on his/her dialing habits and localize it for the outgoing gateway (cd xform patterns), in that way, a US user roaming to UK will be allowed to use the LRG route patterns (using US dialing habits) therefore using local UK resources. The drawback here is that this won't be a cost effective solution, in which case you will have to implement teho/location specific route patterns to route calls out and save costs, Hope this make sense, On Fri, Feb 19, 2010 at 6:23 PM, Jeff Cotter jcot...@voxns.commailto:jcot...@voxns.com wrote: Having trouble understanding how this is supposed to function without Site Specific Route Patterns and Route List/Group. If using only non-site specific route patterns pointed to local route group I see no value in this. Of course I am probably mistaken...hence this message. Scenario below No location specific Route Patterns exist...all route patterns point to local route group. Device roams from HQ to BR1 DMG is US for both. (CSS for both home and Roaming device pool is CSS-LD). Roaming sensitive settings are applied as well as Mobility Settings.which means device will use local route group defined in BRI device pool for all calls based on CSS-LD. User dials 95551212..No problem here since calls will now be sent out BR1-GW as expected with correct digits assuming predot is applied via Called Transformation. However take the same scenario above except the DMG is now changed to UK for roaming phone. Mobility settings are no longer applied meaning CSS does not change. The purpose for this is supposed to be that user does not want to have to dial differently when in a new country. However Local Route Group is still obtained from roaming Device Pool. Since all Route Patterns point to local route group and local route group is now UK ALL calls will now be directed to UK gw. Call will fail as digits PSTN is expecting will be incorrect. What is the excepted way to get calls to NOT route out local gateway but traverse the WAN and go out US gateway. I can't think of how to do this without using location specific Route Patterns and Route Lists which now defeats the purpose of the Local Route Group concept. ARRRH ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Device Mobility with Local Route Groups
Having trouble understanding how this is supposed to function without Site Specific Route Patterns and Route List/Group. If using only non-site specific route patterns pointed to local route group I see no value in this. Of course I am probably mistaken...hence this message. Scenario below No location specific Route Patterns exist...all route patterns point to local route group. Device roams from HQ to BR1 DMG is US for both. (CSS for both home and Roaming device pool is CSS-LD). Roaming sensitive settings are applied as well as Mobility Settings.which means device will use local route group defined in BRI device pool for all calls based on CSS-LD. User dials 95551212..No problem here since calls will now be sent out BR1-GW as expected with correct digits assuming predot is applied via Called Transformation. However take the same scenario above except the DMG is now changed to UK for roaming phone. Mobility settings are no longer applied meaning CSS does not change. The purpose for this is supposed to be that user does not want to have to dial differently when in a new country. However Local Route Group is still obtained from roaming Device Pool. Since all Route Patterns point to local route group and local route group is now UK ALL calls will now be directed to UK gw. Call will fail as digits PSTN is expecting will be incorrect. What is the excepted way to get calls to NOT route out local gateway but traverse the WAN and go out US gateway. I can't think of how to do this without using location specific Route Patterns and Route Lists which now defeats the purpose of the Local Route Group concept. ARRRH ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] MGCP Gateway Problem
Are you using the Bind Media and Bind Control statements on the MGCP gateway? If so make sure the gateway shows registered with that IP address in UCM. If it is registered with a different address remove the bind statements from the gateway and do no mgcp, mgcp. Hth Jeff -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Sunday, February 07, 2010 12:49 PM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 48, Issue 36 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: MGCP Gateway Problem (Jeff Price (jeffpric)) 2. Re: existing VM Ware CLI (Jason Granat) -- Message: 1 Date: Sun, 7 Feb 2010 14:40:51 -0600 From: Jeff Price (jeffpric) jeffp...@cisco.com Subject: Re: [OSL | CCIE_Voice] MGCP Gateway Problem To: afatsum afat...@verizon.net Cc: ccie_voice@onlinestudylist.com Message-ID: b2de0afa86565c47bd3a8435550f95535da...@xmb-rcd-201.cisco.com Content-Type: text/plain; charset=us-ascii It appears that my transformation is working. When dialing 91408425, the display on the phone says To 408425. And the DNA analysis shows what the CUCM is going through process-wise. Yet I am still not receiving any ISDN Q931 debug output on R1 and the phones still receive a fast busy. As I had said in a previous email, the PSTN router that the phones are calling to is already pre-configured and I don't have access to it. Even if it was the PSTN router causing the problem, wouldn't I still see the Q931 output on R1? Thanks for the help. Jeff -Original Message- From: Jeff Price (jeffpric) Sent: Sunday, February 07, 2010 12:35 PM To: 'afatsum' Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] MGCP Gateway Problem I have deactivated all of the services on the SUB and let everything register with the PUB. Jeff -Original Message- From: afatsum [mailto:afat...@verizon.net] Sent: Sunday, February 07, 2010 12:05 PM To: Jeff Price (jeffpric) Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MGCP Gateway Problem Hi Jeff, Can you shutdown the sub and let everything register to pub and then do the testing? This way atleast we can eliminate the sub for troubleshooting purposes. -- Mustafa Jeff Price (jeffpric) wrote: Hi again, I was able to access the DNA on SUB, but not the PUB even though both servers are running the service. Here is the output of DNA. Everything seems to be okay. *Cisco Unified Communications Manager Dialed Number Analyzer Results * * *Results Summary* o *Calling Party Information* + *Calling Party* = +14085252001 + *Partition* = PT_HQ_DEVICES + *Device CSS* = CSS_HQ_DEVICES + *Line CSS* = + *AAR Group Name* = + *AAR CSS* = o *Dialed Digits* = 91408425 o *Match Result* = RouteThisPattern o *Matched Pattern Information* + *Pattern* = \+! + *Partition* = PT_GLOBAL + *Time Schedule* = o *Called Party Number* = +1408425 o *Time Zone* = Pacific Standard/Daylight Time o *End Device* = RL_LOCAL o *Call Classification* = OffNet o *InterDigit Timeout* = NO o *Device Override* = Disabled o *Outside Dial Tone* = NO * *Call Flow* o *TranslationPattern* :*Pattern*= 9.1[2-9]XX[2-9]XX + *Positional Match List* = +1408425 + *Calling Party Number* = +14085252001 + *PreTransform Calling Party Number* = 2001 + *PreTransform Called Party Number* = 91408425 + *Calling Party Transformations* # *External Phone Number Mask* = YES # *Calling Party Mask* = # *Prefix* = # *CallingLineId Presentation* = Default # *CallingName Presentation* = Default # *Calling Party Number* = +14085252001 + *ConnectedParty Transformations* # *ConnectedLineId Presentation* = Default # *ConnectedName Presentation* = Default + *Called Party Transformations*
Re: [OSL | CCIE_Voice] Use voice translation-rule to reject outgoing calls
You could just point the dial-peer for that number to a fictitious port on the router. The call then fail. Hth Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Real world infrastructure question
Curious how the following scenario would be handled in the real world for a UC deployment. Assume the Access switches are layer 2 only. ( Data Vlan 10 Voice Vlan 20) Access switch is trunked to a layer 3 Distribution/Core Switch. Call Manager and Unity are configured in Vlan 30. The phones in Voice Vlan 20 need to reach Call Manager in Vlan 30. Typically there would be SVI interfaces configured on Distribution Switch for each Vlan to handle routing between the Vlans. The problem is... there are no IP Phones/hosts plugged into the Distribution Core Switch in VLAN 20 so the SVI interface will not be in a UP state.therefore no routing between Vlan 20 and Vlan 30 is possible. The only solution I see is to plug an IP Phone into the Distribution/Core switch configured in Vlan 20.I could then take my entire voice network down by unplugging that single phone There must be a better way...what am I missing? Thanks Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Real world infrastructure question
Thanks, I did not realize that merely allowing a vlan across a trunk would enable the SVI. -Original Message- From: Jason Granat [mailto:j...@slash128.com] Sent: Saturday, January 09, 2010 11:05 AM To: Jeff Cotter Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Real world infrastructure question If VLAN 20 is trunked somewhere the SVI will be up, regardless of whether or not there is an access port up in VLAN 20. Having VLAN 20 trunked from dist to access will satisfy this. Sent while mobile. On Jan 9, 2010, at 10:57, Jeff Cotter jcot...@voxns.com wrote: Curious how the following scenario would be handled in the real world for a UC deployment. Assume the Access switches are layer 2 only. ( Data Vlan 10 Voice Vlan 20) Access switch is trunked to a layer 3 Distribution/Core Switch. Call Manager and Unity are configured in Vlan 30. The phones in Voice Vlan 20 need to reach Call Manager in Vlan 30. Typically there would be SVI interfaces configured on Distribution Switch for each Vlan to handle routing between the Vlans. The problem is… there are no IP Phones/hosts plugged into the Distri bution Core Switch in VLAN 20 so the SVI interface will not be in a UP state…..therefore no routing between Vlan 20 and Vlan 30 is possi ble. The only solution I see is to plug an IP Phone into the Distr ibution/Core switch configured in Vlan 20…..I could then take my ent ire voice network down by unplugging that single phone There mus t be a better way…what am I missing? Thanks Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://slash128.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Trying to disable PC Port on CME
Thanks Otto, that fixed it. I read that information 3 times on the cnf-file perphone option will not work unless you change the cnf file location to other place different to system in the admin guide but still could not seem to understand what it was telling me.arrrgg. Thanks again makes sense now. Jeff From: Otto Sanchez [mailto:o...@ipexpert.com] Sent: Wednesday, December 30, 2009 10:25 AM To: Jeff Cotter Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Trying to disable PC Port on CME Hi, (default), try cnf-file location flash: in telephony service configuration, After that you should see if the specific phone configuration file was indeed created, Ah, also configure the phone type in the ephone configuration, HTH On Wed, Dec 30, 2009 at 1:03 PM, Jeff Cotter jcot...@voxns.commailto:jcot...@voxns.com wrote: Can't seem to get this to work. When I go to the phone settings...Device Settings/Security it shows PC Port Disable No. I issue the cnf-file perphone, create cnf-file and reset the phone... Also reloaded router to no avail! . Configuration below. ephone 2 mac-address 0023.EB53.2544 ephone-template 2 button 1:2 ephone-template 2 service phone pcPort 1 service phone settingsAccess 0 ! telephony-service max-ephones 5 max-dn 10 no-reg ip source-address 192.168.1.7 port 2000 caller-id block code *69 cnf-file perphone time-zone 5 voicemail 3099 max-conferences 4 gain -6 call-forward pattern .T web admin system name admin password cisco dn-webedit time-webedit transfer-system full-consult transfer-pattern .T after-hours block pattern 1 1900 7-24 create cnf-files version-stamp Jan 01 2002 00:00:00 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Trying to disable PC Port on CME
Sure Anil, pertinent configuration below. Key commands were under telephony service…I had to have BOTH cnf-file location flash and cnf-file perphone configured. telephony-service max-ephones 5 max-dn 10 no-reg ip source-address 192.168.1.7 port 2000 caller-id block code *69 cnf-file location flash: cnf-file perphone time-zone 5 voicemail 3099 max-conferences 4 gain -6 call-forward pattern .T web admin system name admin password cisco dn-webedit time-webedit transfer-system full-consult transfer-pattern .T after-hours block pattern 1 1900 7-24 create cnf-files version-stamp Jan 01 2002 00:00:00 ! ephone-template 13 service phone pcPort 1 ephone 2 Device Security Mode: Non-Secure mac-address 0023.EB53.2544 type 7965 button 1:2 keepalive 30 auxiliary 30 multicast-moh max-calls-per-button 8 busy-trigger-per-button 0 ephone-template 13 Always send media packets to this router: No Preferred codec: g711ulaw conference drop-mode never conference add-mode all conference admin: No privacy: Yes privacy button: No user-locale US network-locale US From: anil batra [mailto:anil...@yahoo.com] Sent: Wednesday, December 30, 2009 11:35 AM To: Otto Sanchez; Jeff Cotter Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Trying to disable PC Port on CME Jeff, so you had to give cnf-file location flash: under telephone-service and that fixed it. Could you please post your final telephone-service configs. Appreciate it !!! -Anil --- On Thu, 12/31/09, Jeff Cotter jcot...@voxns.com wrote: From: Jeff Cotter jcot...@voxns.com Subject: Re: [OSL | CCIE_Voice] Trying to disable PC Port on CME To: Otto Sanchez o...@ipexpert.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Thursday, December 31, 2009, 12:56 AM Thanks Otto, that fixed it. I read that information 3 times on the “cnf-file perphone option will not work unless you change the cnf file location to other place different to system” in the admin guide but still could not seem to understand what it was telling me…..arrrgg. Thanks again makes sense now. Jeff From: Otto Sanchez [mailto:o...@ipexpert.com] Sent: Wednesday, December 30, 2009 10:25 AM To: Jeff Cotter Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Trying to disable PC Port on CME Hi, (default), try cnf-file location flash: in telephony service configuration, After that you should see if the specific phone configuration file was indeed created, Ah, also configure the phone type in the ephone configuration, HTH On Wed, Dec 30, 2009 at 1:03 PM, Jeff Cotter jcot...@voxns.comhttp://us.mc379.mail.yahoo.com/mc/compose?to=jcot...@voxns.com wrote: Can’t seem to get this to work. When I go to the phone settings…Device Settings/Security it shows PC Port Disable No. I issue the cnf-file perphone, create cnf-file and reset the phone… Also reloaded router to no avail! . Configuration below. ephone 2 mac-address 0023.EB53.2544 ephone-template 2 button 1:2 ephone-template 2 service phone pcPort 1 service phone settingsAccess 0 ! telephony-service max-ephones 5 max-dn 10 no-reg ip source-address 192.168.1.7 port 2000 caller-id block code *69 cnf-file perphone time-zone 5 voicemail 3099 max-conferences 4 gain -6 call-forward pattern .T web admin system name admin password cisco dn-webedit time-webedit transfer-system full-consult transfer-pattern .T after-hours block pattern 1 1900 7-24 create cnf-files version-stamp Jan 01 2002 00:00:00 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.comhttp://www.ipexpert.com/ -Inline Attachment Follows- ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Trying to disable PC Port on CME
No… default location is SYSTEM…I agree, a little confusing especially since the Admin Guide does not address the commands needed under Telephony Service. Thanks again Otto! Jeff From: anil batra [mailto:anil...@yahoo.com] Sent: Wednesday, December 30, 2009 11:46 AM To: Otto Sanchez; Jeff Cotter Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Trying to disable PC Port on CME Awesome !!! Thanks a million,Jeff. A little confusion here...is not cnf-file location flash: the default location for phone config files ??? --- On Thu, 12/31/09, Jeff Cotter jcot...@voxns.com wrote: From: Jeff Cotter jcot...@voxns.com Subject: Re: [OSL | CCIE_Voice] Trying to disable PC Port on CME To: anil batra anil...@yahoo.com, Otto Sanchez o...@ipexpert.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Thursday, December 31, 2009, 1:12 AM Sure Anil, pertinent configuration below. Key commands were under telephony service…I had to have BOTH cnf-file location flash and cnf-file perphone configured. telephony-service max-ephones 5 max-dn 10 no-reg ip source-address 192.168.1.7 port 2000 caller-id block code *69 cnf-file location flash: cnf-file perphone time-zone 5 voicemail 3099 max-conferences 4 gain -6 call-forward pattern .T web admin system name admin password cisco dn-webedit time-webedit transfer-system full-consult transfer-pattern .T after-hours block pattern 1 1900 7-24 create cnf-files version-stamp Jan 01 2002 00:00:00 ! ephone-template 13 service phone pcPort 1 ephone 2 Device Security Mode: Non-Secure mac-address 0023.EB53.2544 type 7965 button 1:2 keepalive 30 auxiliary 30 multicast-moh max-calls-per-button 8 busy-trigger-per-button 0 ephone-template 13 Always send media packets to this router: No Preferred codec: g711ulaw conference drop-mode never conference add-mode all conference admin: No privacy: Yes privacy button: No user-locale US network-locale US From: anil batra [mailto:anil...@yahoo.com] Sent: Wednesday, December 30, 2009 11:35 AM To: Otto Sanchez; Jeff Cotter Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Trying to disable PC Port on CME Jeff, so you had to give cnf-file location flash: under telephone-service and that fixed it. Could you please post your final telephone-service configs. Appreciate it !!! -Anil --- On Thu, 12/31/09, Jeff Cotter jcot...@voxns.com wrote: From: Jeff Cotter jcot...@voxns.com Subject: Re: [OSL | CCIE_Voice] Trying to disable PC Port on CME To: Otto Sanchez o...@ipexpert.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Thursday, December 31, 2009, 12:56 AM Thanks Otto, that fixed it. I read that information 3 times on the “cnf-file perphone option will not work unless you change the cnf file location to other place different to system” in the admin guide but still could not seem to understand what it was telling me…..arrrgg. Thanks again makes sense now. Jeff From: Otto Sanchez [mailto:o...@ipexpert.com] Sent: Wednesday, December 30, 2009 10:25 AM To: Jeff Cotter Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Trying to disable PC Port on CME Hi, (default), try cnf-file location flash: in telephony service configuration, After that you should see if the specific phone configuration file was indeed created, Ah, also configure the phone type in the ephone configuration, HTH On Wed, Dec 30, 2009 at 1:03 PM, Jeff Cotter jcot...@voxns.comhttp://us.mc379.mail.yahoo.com/mc/compose?to=jcot...@voxns.com wrote: Can’t seem to get this to work. When I go to the phone settings…Device Settings/Security it shows PC Port Disable No. I issue the cnf-file perphone, create cnf-file and reset the phone… Also reloaded router to no avail! . Configuration below. ephone 2 mac-address 0023.EB53.2544 ephone-template 2 button 1:2 ephone-template 2 service phone pcPort 1 service phone settingsAccess 0 ! telephony-service max-ephones 5 max-dn 10 no-reg ip source-address 192.168.1.7 port 2000 caller-id block code *69 cnf-file perphone time-zone 5 voicemail 3099 max-conferences 4 gain -6 call-forward pattern .T web admin system name admin password cisco dn-webedit time-webedit transfer-system full-consult transfer-pattern .T after-hours block pattern 1 1900 7-24 create cnf-files version-stamp Jan 01 2002 00:00:00 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.comhttp://www.ipexpert.com/ -Inline Attachment Follows- ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -Inline Attachment Follows- ___ For more information
[OSL | CCIE_Voice] CUE image change to UCM
Could someone post or point me to a post or a document that outlines how to change the CUE image/license to integrate to Call Manager as opposed to CME and vice a versa. Thanks Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Transcoder not engaging
Having problems getting my txcoders to work on new CME. Shows registered but all g729 to g711 calls fail. Configs included. DSP farm shows as registered and enabled. Any help would be appreciated. voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip voice-card 0 dsp services dspfarm sccp local FastEthernet0/0 sccp ccm 192.168.1.7 identifier 1 priority 1 sccp ! sccp ccm group 1 bind interface FastEthernet0/0 associate ccm 1 priority 1 associate profile 1 register localtxc ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 2 associate application SCCP telephony-service sdspfarm units 1 sdspfarm transcode sessions 4 sdspfarm tag 1 localtxc max-ephones 2 max-dn 20 ip source-address 192.168.1.7 port 2000 auto assign 1 to 2 url services http://192.168.1.8/voiceview/common/login.do url authentication http://192.168.1.8/voiceview/authentication/authenticate.do voicemail 3099 max-conferences 4 gain -6 call-forward pattern .T moh music-on-hold.au web admin system name admin password cisco dn-webedit time-webedit transfer-system full-consult transfer-pattern .T create cnf-files version-stamp Jan 01 2002 00:00:00 R2801#sh sccp SCCP Admin State: UP Gateway IP Address: 192.168.1.7, Port Number: 2000 IP Precedence: 5 User Masked Codec list: None Call Manager: 192.168.1.7, Port Number: 2000 Priority: 1, Version: 3.1, Identifier: 1 Transcoding Oper State: ACTIVE - Cause Code: NONE Active Call Manager: 192.168.1.7, Port Number: 2000 TCP Link Status: CONNECTED, Profile Identifier: 1 Reported Max Streams: 4, Reported Max OOS Streams: 0 Supported Codec: g711ulaw, Maximum Packetization Period: 30 Supported Codec: g711alaw, Maximum Packetization Period: 30 Supported Codec: g729ar8, Maximum Packetization Period: 60 Supported Codec: g729abr8, Maximum Packetization Period: 60 Supported Codec: g729r8, Maximum Packetization Period: 60 Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30 Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30 Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period R2801#sh sdspfarm units mtp-1 Device:localtxc TCP socket:[2] REGISTERED in SCCP ver 0/10 actual_stream:4 max_stream 4 IP:192.168.1.7 58193 MTP YOKO keepalive 24 Supported codec: G711Ulaw G711Alaw G729 G729a G729ab max-mtps:1, max-streams:8, alloc-streams:4, act-streams:0 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Transcoder not engaging
Thanks for the replies. DP 1 is the incoming DP I did not include the other with session target of RAS. The problem turned out to be the lack of voice-class codec command on DP 1. Interestingly enough this also broke MW notification!! To summarize- remote site (another CME) has dial-peer 10 session target ras with codec g711 hardcoded in DP. Terminating Site had a DP 1 defined with incoming called number of (.) and NO codec or Voice-Class Codec defined. I assumed it would default to g729 and since there would now be a codec mismatch my transcoder would be invoked. Not the case. As soon as I configured the voice class with codecs g711 and g729 and applied to dial-peer 1 on terminating CME everything started working (Big Thanks to Michael Ciarfello for pointing this out!) including MW notification! I can now hardcode the remote DP to g729 or g711 and the call completes. If I hard code the remote DP to g729 and then make a call and let the call FNA to CUE than my transcoder is invoked. Confirmed all the above with show and debugs. I can also duplicate the problem including breaking the MW by removing the voice-class command. One other point on this is if I remove DP 1 all together and let the Default DP handle the incoming leg everything worksARGH!! Always thought having the default DP involved was a big no no! Thanks again for the replies and support. Jeff From: vccie2010 [mailto:vccie2...@gmail.com] Sent: Wednesday, October 21, 2009 4:26 PM To: Jeff Cotter Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Transcoder not engaging I don't see session target ras on DP voip 1 Not sure how you are still getting calls working. from UCM to CME via GK. On Wed, Oct 21, 2009 at 12:29 PM, Jeff Cotter jcot...@voxns.commailto:jcot...@voxns.com wrote: Having problems getting my txcoders to work on new CME. Shows registered but all g729 to g711 calls fail. Configs included. DSP farm shows as registered and enabled. Any help would be appreciated. voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip voice-card 0 dsp services dspfarm sccp local FastEthernet0/0 sccp ccm 192.168.1.7 identifier 1 priority 1 sccp ! sccp ccm group 1 bind interface FastEthernet0/0 associate ccm 1 priority 1 associate profile 1 register localtxc ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 2 associate application SCCP telephony-service sdspfarm units 1 sdspfarm transcode sessions 4 sdspfarm tag 1 localtxc max-ephones 2 max-dn 20 ip source-address 192.168.1.7 port 2000 auto assign 1 to 2 url services http://192.168.1.8/voiceview/common/login.do url authentication http://192.168.1.8/voiceview/authentication/authenticate.do voicemail 3099 max-conferences 4 gain -6 call-forward pattern .T moh music-on-hold.au web admin system name admin password cisco dn-webedit time-webedit transfer-system full-consult transfer-pattern .T create cnf-files version-stamp Jan 01 2002 00:00:00 R2801#sh sccp SCCP Admin State: UP Gateway IP Address: 192.168.1.7, Port Number: 2000 IP Precedence: 5 User Masked Codec list: None Call Manager: 192.168.1.7, Port Number: 2000 Priority: 1, Version: 3.1, Identifier: 1 Transcoding Oper State: ACTIVE - Cause Code: NONE Active Call Manager: 192.168.1.7, Port Number: 2000 TCP Link Status: CONNECTED, Profile Identifier: 1 Reported Max Streams: 4, Reported Max OOS Streams: 0 Supported Codec: g711ulaw, Maximum Packetization Period: 30 Supported Codec: g711alaw, Maximum Packetization Period: 30 Supported Codec: g729ar8, Maximum Packetization Period: 60 Supported Codec: g729abr8, Maximum Packetization Period: 60 Supported Codec: g729r8, Maximum Packetization Period: 60 Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30 Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30 Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period R2801#sh sdspfarm units mtp-1 Device:localtxc TCP socket:[2] REGISTERED in SCCP ver 0/10 actual_stream:4 max_stream 4 IP:192.168.1.7 58193 MTP YOKO keepalive 24 Supported codec: G711Ulaw G711Alaw G729 G729a G729ab max-mtps:1, max-streams:8, alloc-streams:4, act-streams:0 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Cisco Software Download site.
Going to upgrade the IOS on my new router, Cisco has two packages for each IOS. Can someone explain the difference between The Feature Set Factory Upgrade For Bundles and just the straight IOS feature set? Thanks. Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Cisco Software Download site.
Yes I ordered the 2801 which came with the SP services 12.4.15 T(10) as you mentioned. It also came with CME 4.0.1. I would like to upgrade the IOS as to support CME 7.X and then download and install the correct TAR file for CME 7.X. If I understand you correctly I would choose the Feature Set Factory Upgrade for Bundles for SP service 12.4.22T? Correct. From: Michael Ciarfello [mailto:mciarfe...@iplogic.com] Sent: Monday, October 19, 2009 10:42 AM To: Jeff Cotter; ccie_voice@onlinestudylist.com Subject: RE: Cisco Software Download site. What did you purchase with your router? For example, if you purchased a 2811-CCME bundle, then it comes with SP Services by default (which is usually 12.4.15Tsomething). If at the time of ordering you wanted security, you upgraded the bundle SP Services to Advanced IP Services. In this case, you SHOULD choose factory upgrade for bundles to upgrade your advanced ip from the factory installed 12.4.15 to your required version, say 12.4.22t2. They are the same but one is more correct than the other. Maybe someone else's account would only display the factory upgrade for bundles if that is the only router they ever ordered and is associated with their account. Maybe Cisco is just tracking internally. REMEMBER, that the download page says you may be liable for downloading software you are not licensed for. Never heard of Cisco going after people, but they might start to enforce it some day. TAC is getting MUCH stricter in opening cases. Need the serial number now most of the time. Software downloads might someday too. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Cotter Sent: Monday, October 19, 2009 1:20 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Cisco Software Download site. Going to upgrade the IOS on my new router, Cisco has two packages for each IOS. Can someone explain the difference between The Feature Set Factory Upgrade For Bundles and just the straight IOS feature set? Thanks. Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CME Flash Location
Just received new 2801 with CME/UE. Phones are working fine. Where the heck are the CME files/BACD etc on flash?? When I do a sh flash I see Router#sh flash: -#- --length-- -date/time-- path 1 37224796 Sep 30 2009 16:04:40 +00:00 c2801-spservicesk9-mz.124-15.T10.bin 2 2746 Sep 30 2009 16:27:00 +00:00 sdmconfig-2801.cfg 3 931840 Sep 30 2009 16:27:22 +00:00 es.tar 4 1505280 Sep 30 2009 16:27:46 +00:00 common.tar 5 1038 Sep 30 2009 16:28:08 +00:00 home.shtml 6 112640 Sep 30 2009 16:28:26 +00:00 home.tar 7 1697952 Sep 30 2009 16:28:58 +00:00 securedesktop-ios-3.1.1.45-k9.pkg 8 415956 Sep 30 2009 16:29:24 +00:00 sslclient-win-1.1.4.176.pkg 9 660 Oct 17 2009 20:22:02 +00:00 vlan.dat ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Employment Opportunity Northern California
I know of a company looking for an experienced Cisco voice person for work in Northern California. If there is any interest please contact me directly for details. Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Electronic Readers
Since we have to do so much reading, I was curious if anybody has been using devices such as Kindle or Sony Readers to read Cisco PDF docs and how well it works etc. Thanks Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] suggestion
I learn a great deal from this list and it's participants and appreciate the efforts that are put into keeping it active and up to date, however since were on the subject of making our study list better:) I would like to see questions that are posted which go unanswered or unresolved for 4 or 5 days be addressed directly by the moderator or other IPexpert personnel. I see some posts that are answered directly by Mark, Vic or Wayne immediately, other posts receive no input from members the moderator or other IPexpert personnel? Not sure why that is or what the rules of engagement are. There may be a very good explanation for this, if so please educate me. Even if the response is we don't know or never heard of that would be good. Again, just a suggestion overall a great resource for us wannabees. Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] suggestion
Thanks Wayne, understood and all valid points. Maybe a better way to word my suggestion would be. If and when IPexpert personnel do review the posts and have time to respond (obviously it does happen) they could try and target posts that have gone unresolved for 4 or 5 days. -Original Message- From: Wayne Lawson [mailto:groupst...@ipexpert.com] Sent: Friday, September 18, 2009 2:36 PM To: Jeff Cotter Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] suggestion Jeff - Great suggestion - and if I could make that happen - I would. However, my instructors also teach and update / create products. If everyone on the list - and the other clients are okay with products taking twice as long to get to market - I can ask my developers to stop what they're doing and answer support questions.(sarcastic). The reason for the list is for group participation. If there's something that's not explained in our Solution Guide - email me directly and I can get an instructor to look into it. If it's a matter of simply not understanding the technologies or theory - that's what our classes are there for. Regards, Wayne A. Lawson II - CCIE #5244 Founder President - IPexpert, Inc. Mailto: wlaw...@ipexpert.com Mobile: +1.810.334.1564 :: Message sent from iPhone. On Sep 18, 2009, at 5:29 PM, Jeff Cotter jcot...@voxns.com wrote: I learn a great deal from this list and it’s participants and apprec iate the efforts that are put into keeping it active and up to date, however since were on the subject of making our study list better…. :) I would like to see questions that are posted which go unanswere d or unresolved for 4 or 5 days be addressed directly by the moderat or or other IPexpert personnel. I see some posts that are answered directly by Mark, Vic or Wayne immediately, other posts receive no input from members the moderator or other IPexpert personnel? Not s ure why that is or what the “rules of engagement” are. There may be a very good explanation for this, if so please educate me. Even if the response is “we don’t know” or “never heard of that” would be good. Again, just a suggestion overall a great resou rce for us “wannabees”. Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Link Loss Type explanation
Looking for an explanation or a good document that explains this setting under Region configuration. Can someone point me in the right direction on this. Thanks Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] IPCC Administration issue
OK, back in business. I had to perform the password recovery procedure outlined below. Thanks for the responses. Still not sure what caused this!! * On versions 5, 6 and 7: 1) Go to Start, run, type 'cet' on the UCCX Server. This will launch the Configuration Object Editor. 2) Browse to: com.cisco.crs.cluster.config.AppAdminSetupConfig in the left hand pane. 3) Right click the row on the right and hit modify. Then select the 'com.cisco.crs.cluster.config.AppAdminSetupConfig' tab. 4) Change the setup state to: FRESH_INSTALL and hit OK 5) Log into the CRA App Admin page with the default username (may be case sensitive): Administrator and password: ciscocisco From: Tanner Ezell [mailto:tanner.ez...@gmail.com] Sent: Wednesday, August 12, 2009 1:21 PM To: Jeff Cotter Subject: Re: [OSL | CCIE_Voice] IPCC Administration issue Are you able to navigate to http://localhost/appadmin/RmCm?request_type=rmjtapi.configure ? On Wed, Aug 12, 2009 at 1:19 PM, Jeff Cotter jcot...@voxns.commailto:jcot...@voxns.com wrote: Well successful as in I get to the main page however I am unable to do anything once there. When I click on System or Tools it does not do anything. No drop down menu of choices etc. From: Tanner Ezell [mailto:tanner.ez...@gmail.commailto:tanner.ez...@gmail.com] Sent: Wednesday, August 12, 2009 1:17 PM To: Jeff Cotter Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IPCC Administration issue That's very odd. You did get a successful login with Administrator/ciscocisco right? On Wed, Aug 12, 2009 at 12:35 PM, Jeff Cotter jcot...@voxns.commailto:jcot...@voxns.com wrote: No luck. I cleared cache. Browse to 10.1.100.13/appadminhttp://10.1.100.13/appadmin login with Administrator/ciscocisco. Then browse to http://localhost/appadmin/JTAPI logged in user not authorized to view this page From: Tanner Ezell [mailto:tanner.ez...@gmail.commailto:tanner.ez...@gmail.com] Sent: Wednesday, August 12, 2009 12:18 PM To: Jeff Cotter Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IPCC Administration issue Clear cache and try again. Login first through appadmin with Administrator/ciscocisco, then navigate to that page. It'll work, I've had to do the same before. On Wed, Aug 12, 2009 at 12:14 PM, Jeff Cotter jcot...@voxns.commailto:jcot...@voxns.com wrote: Thanks Tanner, when I go to http://localhost/appadmin/JTAPI I am prompted for login info. Which the only thing that works is Administrator/ciscocisco it then tells me the logged in user is unauthorized to access this page. As such there is no option for me to run the JTAPI Resync. From: Tanner Ezell [mailto:tanner.ez...@gmail.commailto:tanner.ez...@gmail.com] Sent: Wednesday, August 12, 2009 12:02 PM To: Jeff Cotter Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IPCC Administration issue Navigate to http://localhost/appadmin/JTAPI and run the JTAPI Resync On Wed, Aug 12, 2009 at 11:32 AM, Jeff Cotter jcot...@voxns.commailto:jcot...@voxns.com wrote: I booted my IPCC server for the first time in awhile and had problems logging in to app admin. The old password I used was not working. If I use the default password Administrator/ciscocisco...I am authenticated however when I select System or Tools nothing happens. I do not get the drop down menus and therefore am unable to do anything. In addition my Auto Attendant I have configured on IPCC is working fine. The only changes I can remember making that may be affecting this are the following. LDAP integration. My crsadmin user was pending deletion. I removed ldap and rebooted. Same issue. I changed the Name of my sever to an IP address however I could swear I tested everything after that I am admittedly very weak with Microsoft ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] IPCC Administration issue
I booted my IPCC server for the first time in awhile and had problems logging in to app admin. The old password I used was not working. If I use the default password Administrator/ciscocisco...I am authenticated however when I select System or Tools nothing happens. I do not get the drop down menus and therefore am unable to do anything. In addition my Auto Attendant I have configured on IPCC is working fine. The only changes I can remember making that may be affecting this are the following. LDAP integration. My crsadmin user was pending deletion. I removed ldap and rebooted. Same issue. I changed the Name of my sever to an IP address however I could swear I tested everything after that I am admittedly very weak with Microsoft ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] IPCC Administration issue
No luck. I cleared cache. Browse to 10.1.100.13/appadmin login with Administrator/ciscocisco. Then browse to http://localhost/appadmin/JTAPI logged in user not authorized to view this page From: Tanner Ezell [mailto:tanner.ez...@gmail.com] Sent: Wednesday, August 12, 2009 12:18 PM To: Jeff Cotter Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IPCC Administration issue Clear cache and try again. Login first through appadmin with Administrator/ciscocisco, then navigate to that page. It'll work, I've had to do the same before. On Wed, Aug 12, 2009 at 12:14 PM, Jeff Cotter jcot...@voxns.commailto:jcot...@voxns.com wrote: Thanks Tanner, when I go to http://localhost/appadmin/JTAPI I am prompted for login info. Which the only thing that works is Administrator/ciscocisco it then tells me the logged in user is unauthorized to access this page. As such there is no option for me to run the JTAPI Resync. From: Tanner Ezell [mailto:tanner.ez...@gmail.commailto:tanner.ez...@gmail.com] Sent: Wednesday, August 12, 2009 12:02 PM To: Jeff Cotter Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IPCC Administration issue Navigate to http://localhost/appadmin/JTAPI and run the JTAPI Resync On Wed, Aug 12, 2009 at 11:32 AM, Jeff Cotter jcot...@voxns.commailto:jcot...@voxns.com wrote: I booted my IPCC server for the first time in awhile and had problems logging in to app admin. The old password I used was not working. If I use the default password Administrator/ciscocisco...I am authenticated however when I select System or Tools nothing happens. I do not get the drop down menus and therefore am unable to do anything. In addition my Auto Attendant I have configured on IPCC is working fine. The only changes I can remember making that may be affecting this are the following. LDAP integration. My crsadmin user was pending deletion. I removed ldap and rebooted. Same issue. I changed the Name of my sever to an IP address however I could swear I tested everything after that I am admittedly very weak with Microsoft ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] IPCC Administration issue
Well successful as in I get to the main page however I am unable to do anything once there. When I click on System or Tools it does not do anything. No drop down menu of choices etc. From: Tanner Ezell [mailto:tanner.ez...@gmail.com] Sent: Wednesday, August 12, 2009 1:17 PM To: Jeff Cotter Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IPCC Administration issue That's very odd. You did get a successful login with Administrator/ciscocisco right? On Wed, Aug 12, 2009 at 12:35 PM, Jeff Cotter jcot...@voxns.commailto:jcot...@voxns.com wrote: No luck. I cleared cache. Browse to 10.1.100.13/appadminhttp://10.1.100.13/appadmin login with Administrator/ciscocisco. Then browse to http://localhost/appadmin/JTAPI logged in user not authorized to view this page From: Tanner Ezell [mailto:tanner.ez...@gmail.commailto:tanner.ez...@gmail.com] Sent: Wednesday, August 12, 2009 12:18 PM To: Jeff Cotter Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IPCC Administration issue Clear cache and try again. Login first through appadmin with Administrator/ciscocisco, then navigate to that page. It'll work, I've had to do the same before. On Wed, Aug 12, 2009 at 12:14 PM, Jeff Cotter jcot...@voxns.commailto:jcot...@voxns.com wrote: Thanks Tanner, when I go to http://localhost/appadmin/JTAPI I am prompted for login info. Which the only thing that works is Administrator/ciscocisco it then tells me the logged in user is unauthorized to access this page. As such there is no option for me to run the JTAPI Resync. From: Tanner Ezell [mailto:tanner.ez...@gmail.commailto:tanner.ez...@gmail.com] Sent: Wednesday, August 12, 2009 12:02 PM To: Jeff Cotter Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IPCC Administration issue Navigate to http://localhost/appadmin/JTAPI and run the JTAPI Resync On Wed, Aug 12, 2009 at 11:32 AM, Jeff Cotter jcot...@voxns.commailto:jcot...@voxns.com wrote: I booted my IPCC server for the first time in awhile and had problems logging in to app admin. The old password I used was not working. If I use the default password Administrator/ciscocisco...I am authenticated however when I select System or Tools nothing happens. I do not get the drop down menus and therefore am unable to do anything. In addition my Auto Attendant I have configured on IPCC is working fine. The only changes I can remember making that may be affecting this are the following. LDAP integration. My crsadmin user was pending deletion. I removed ldap and rebooted. Same issue. I changed the Name of my sever to an IP address however I could swear I tested everything after that I am admittedly very weak with Microsoft ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Presence integration issue
Aamir, the only way I was able to get presence/IM working was to setup AD/DNS. I was never able to get it working without. I also found a pretty good book on CUPS which has some really good troubleshooting tips. BTW, this book says CUP without AD is not recommended. Does not say it won't work just not recommended. Title is Deploying Cisco Unified Presence author is HouTong Luo, CCIE no. 6183 available at Amazon. From: Aamir Panjwani [mailto:aamir.panjw...@ivision.com.au] Sent: Thursday, July 30, 2009 11:25 PM To: Jeff Cotter; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Presence integration issue Hi Guys, I am also running into exact same problem Jeff mentioned below. Everything works except presence/IM. I would be keen to know if someone else managed to have a fully functional CUPC going without AD/DNS. Can some please point me in the right direction Thanks From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Cotter Sent: Sunday, 14 June 2009 9:54 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Presence integration issue I appear to have everything working EXCEPT presence. Personal Communicator is showing limited connectivity...no presence information..indicates off line and grayed outI can make and receive calls, view voicemail and use IP phone messenger. Troubleshooting window indicates all green checkmarks Based on a Cisco troubleshooting document I found, I am being pointed to the Proxy Domain name under SystemService Parameters my cups serverCisco UP Sip Proxy. I am not sure what to enter here as I do not seem to have a domain name configured on either CUCM or CUPS (at least that I can find). Any help would be appreciated. Thanks __ 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 __ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] LDAP Synchronization with Call Manager
Can someone please educate me on the proper entry for ldap user search space. On my DC I have created a brand new OU named cisco, added users to it. My entry in call manager for ldap user search space is ou=cisco, dc=lab. However when I try to sync it never seems to finish. The cancel sync process is my only option. Cannot seem to find a good explanation of this anywhere. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] LDAP Synchronization with Call Manager
Thanks for the reply.. it is lab.com. I made the change however same result. From: Steve Sarrick [mailto:ssarr...@drsllc.net] Sent: Friday, July 10, 2009 11:47 AM To: Jeff Cotter; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] LDAP Synchronization with Call Manager If your fqdn for the domain is lab.com than this should be ou=cisco, dc=lab, dc=com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Cotter Sent: Friday, July 10, 2009 2:41 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] LDAP Synchronization with Call Manager Can someone please educate me on the proper entry for ldap user search space. On my DC I have created a brand new OU named cisco, added users to it. My entry in call manager for ldap user search space is ou=cisco, dc=lab. However when I try to sync it never seems to finish. The cancel sync process is my only option. Cannot seem to find a good explanation of this anywhere. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] LDAP Synchronization with Call Manager
Steve, my mistake that did seem to fix the problem. The sync never indicates that is finishes however it did import all the users. Thanks very much for the help. From: Steve Sarrick [mailto:ssarr...@drsllc.net] Sent: Friday, July 10, 2009 11:47 AM To: Jeff Cotter; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] LDAP Synchronization with Call Manager If your fqdn for the domain is lab.com than this should be ou=cisco, dc=lab, dc=com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Cotter Sent: Friday, July 10, 2009 2:41 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] LDAP Synchronization with Call Manager Can someone please educate me on the proper entry for ldap user search space. On my DC I have created a brand new OU named cisco, added users to it. My entry in call manager for ldap user search space is ou=cisco, dc=lab. However when I try to sync it never seems to finish. The cancel sync process is my only option. Cannot seem to find a good explanation of this anywhere. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Presence integration issue
I appear to have everything working EXCEPT presence. Personal Communicator is showing limited connectivity.no presence information..indicates off line and grayed out..I can make and receive calls, view voicemail and use IP phone messenger. Troubleshooting window indicates all green checkmarks Based on a Cisco troubleshooting document I found, I am being pointed to the Proxy Domain name under SystemService Parameters my cups serverCisco UP Sip ProxyProxy Domain. I am not sure what to enter here as I do not seem to have a domain name configured on either CUCM or CUPS (at least that I can find). Any help would be appreciated. Thanks I appear to be having some trouble posting so I apologize in advance if multiple copies of this are posted. Jeff
[OSL | CCIE_Voice] CUPS Integration issues
I appear to have everything working EXCEPT presence. Personal Communicator is showing limited connectivity.no presence information indicates off line and grayed out..I can make and receive calls, view voicemail and use IP phone messenger. Troubleshooting window indicates all green checkmarks Based on a Cisco troubleshooting document I found, I am being pointed to the Proxy Domain name under SystemService Parameters my cups serverCisco UP Sip Proxy. I am not sure what to put here as I do not seem to have a domain name configured on either CUCM or CUPS Maybe I am to frazzled to find it. Any help would be appreciated. Thanks
[OSL | CCIE_Voice] Presence integration issue
I appear to have everything working EXCEPT presence. Personal Communicator is showing limited connectivity.no presence information..indicates off line and grayed out..I can make and receive calls, view voicemail and use IP phone messenger. Troubleshooting window indicates all green checkmarks Based on a Cisco troubleshooting document I found, I am being pointed to the Proxy Domain name under SystemService Parameters my cups serverCisco UP Sip Proxy. I am not sure what to enter here as I do not seem to have a domain name configured on either CUCM or CUPS (at least that I can find). Any help would be appreciated. Thanks
Re: [OSL | CCIE_Voice] XML PARSE ERROR
Yes, thank you for asking. I finally found out 7940-7960 SIP does not support Phone Messenger. See link below. http://www.cisco.com/en/US/docs/voice_ip_comm/cups/7_0/english/compatibility /cupcompatibility7x.html From: Michael Ciarfello [mailto:mciarfe...@iplogic.com] Sent: Thursday, June 11, 2009 10:03 PM To: Jeff Cotter; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] XML PARSE ERROR Did you resolve this? I get the same thing on an 7940/60 with SIP 8-8-0. Any service including corporate directory gives the error. A 70, 41, etc works. The kind of interesting thing is the 7940/60 end-user phone guide doesn't say Services and Services Button works for SIP, but Corporate directory says it's supported, but still get the XML error. Very confused. _ From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Cotter [jcot...@voxns.com] Sent: Wednesday, June 10, 2009 3:54 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] XML PARSE ERROR I am receiving the following error when trying to access the IP Phone Messenger Service. ( XML PARSE ERROR) from 7960 SIP phone. I believe it may have to do with soft keys. I see a similar bug on Cisco for accessing Corporate Directory but this should have been fixed in updated firmware. Phone is running Sip Firmware 8-12-0. I have also tried 8-9-0 and 8-8-0 with the same results. Call Manager and CUP are both on 7.0. Works fine with SCCP phones.Thank you. Jeff Cotter Director of Engineering/IT Vox Network Solutions 650-989-1021 www.voxns.com
[OSL | CCIE_Voice] XML PARSE ERROR
I am receiving the following error when trying to access the IP Phone Messenger Service. ( XML PARSE ERROR) from 7960 SIP phone. I believe it may have to do with soft keys. I see a similar bug on Cisco for accessing Corporate Directory but this should have been fixed in updated firmware. Phone is running Sip Firmware 8-12-0. I have also tried 8-9-0 and 8-8-0 with the same results. Call Manager and CUP are both on 7.0. Works fine with SCCP phones.Thank you. Jeff Cotter Director of Engineering/IT Vox Network Solutions 650-989-1021 www.voxns.com
