Re: [OSL | CCIE_Voice] Transcoder AGAIN!
Jeff, I believe this is a pretty well known bug :) Chad On Mon, Apr 5, 2010 at 10:47 AM, Jeff Cotter jcot...@voxns.com wrote: 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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCCX script question
They are populated while ringing and they are only called ecc's when
using cad with ucce not uccx. I think tanner hit the nail on the head
though with the layout and set enterprise data steps.
Chad Stachowicz
415-794-8770
Please excuse any mispellings as this message was from my mobile device
On Mar 25, 2010, at 9:22 AM, Tanner Ezell tanner.ez...@gmail.com
wrote:
I'd have to check, but I don't believe the ECC fields are populated in
CAD while the phone is ringing. Only after.
Aside from that, you must use the Set Enterprise Info step, make sure
the field is added to the default layout of the agent and you'll be
set.
---
I just re-read what you posted, and it sounds like you created another
layout list, if that is correct you must specify it with the Set
Enterprise Info step, or add it to the default layout.
On Thu, Mar 25, 2010 at 2:16 PM, Cristobal Priego
cristobalpri...@gmail.com wrote:
Hello Tanner,
I have a question. I know I'm doing something wrong. when the call
is sent
to the agent. on the Data Field part. I see ANI, DNIS, Layout
(default)
what I'm trying to do is, based on the called number i will make a
comparison in a loop, when i find a match in the script based on
the called
number, i want to set a variable with a name. then i would like the
agent to
be able to see ANI, DNIS and the customized name on the Data Field.
i'd like to do something like this
Called Number = 2003
if (DNIS == 2001 )
{
Customer = Safeway
}
else if (DNIS== 2002)
{
Customer = Raleys
}
else if (DNIS == 2003)
{
Customer = SafeMart
}
then I'd like the agent to see on the Data field when the phone is
ringing
on the agent
ANI = 408-123-4567
DNIS = 2003
Customer = SafeMart
I don't know or I can't get the last part to work
I have enhanced license on my uccx version 5.0.2.
on the desktop administrator under Enterprise Data Configuration
I created my field list:
0 customer
and layout list:
Customer : ANI, DNIS, Customer, user.layout
i went to workflow groups agents key accounts work flow,
on the event Ringing Action I only have these options: Run Macro,
Call
Control, Launch External App, Agent state, Utility Action
and i don't know what to do
how do i get this to work ?
thanks
2010/3/25 Tanner Ezell tanner.ez...@gmail.com
The document was sent to the list, but apparently has yet to be
approved (apparently no one pays attention?).
The document can be found here:
http://tannerezell.com/media/UCCX%20Custom%20Reports%20-%20Scripting%20-%20Get%20Session%20ID.pdf
As an aside, you may want to consider the IP of the person who
created
the document before you start passing it around :)
Cheers
On Thu, Mar 25, 2010 at 11:26 AM, Cristobal Priego
cristobalpri...@gmail.com wrote:
No Randall of course i have no problem to share it with the
community. i
will do so
2010/3/24 Randall Saborio ill2...@gmail.com
Lucky Cristobal.
Are you concerned about your intellectual property rights, or
will you
share it with all of us? :)
We won't get mad if you want to keep it private, but just
suggesting.
On Wed, Mar 24, 2010 at 12:03 PM, Cristobal Priego
cristobalpri...@gmail.com wrote:
thank you very much
2010/3/24 Tanner Ezell tanner.ez...@gmail.com
Eh?
I've attached a document I developed which explains everything
you
need to get information from the script to the CAD software.
On Wed, Mar 24, 2010 at 12:29 PM, Cristobal Priego
cristobalpri...@gmail.com wrote:
Hello Tanner,
thanks for your reply.
I have a question, I'm trying to do my customized Get call
Contact
Info
Step
with the Set Enterprise call info step , none of the options
that
i
created
on my enterprise data configuration shows up
i know I'm not understanding something properly. Can I do a Get
call
Contact
Info Step and push some personalized (customized) Strings to
the
Agent
desktop ?
am I on the right track ?
thanks
2010/3/23 Tanner Ezell tanner.ez...@gmail.com
Add a new variable to the work flow
Use the set enterprise call info step to pass the variable
along.
On Tue, Mar 23, 2010 at 6:26 PM, Cristobal Priego
cristobalpri...@gmail.com wrote:
Hello,
I'd like to get some advice on this.
I need to create a script that will get some variables from
the
customer.
and I'd like those variables to be displayed on the Agent
Desktop.
I've been looking on Enterprise Data Format and on the
Desktop
Administrator
I created a workflow and I Modified a few fields for
Enterprise
Data
and
Added a new layout list.
Where i'm having problems is at the time to put all of this
together.
could
you please help me
which steps do i need to use
thanks
___
For more information regarding industry leading CCIE Lab
training,
please
visit www.ipexpert.com
--
Cheers,
Tanner Ezell
--
Cheers
[OSL | CCIE_Voice] Help with IPExpert Material
I'm currently just starting my firswt CCIE V3 blueprint session, and I have my proctor guide and workbook, and I am familiar with IP Expert and can log into device and such. However the workbook does not have the IP Address and VLan's for each POD laid out anywhere nor do i see it anywhere in the proctor labs itself. It says in the workbook to get it from the configuration files on pexpert.com, however when i try to download the CCIE V3 Configuration files, it says it is unable to process that request. Any ideas?! Thanks! Chad ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] ccie voice stats
I think in general its a bit crazy that people think having a CCIE makes a difference in anything. a CCIE only matters if you are new to a company who wnats your cert, or your verbal and 'prior works' aren't engouh to speak for themselves. I can say the best engineers making the most money I know do not have their CCIE's. Just a thought... forgot to reply to all :) Chad On Fri, Oct 30, 2009 at 12:37 AM, Jonathan Charles jonv...@gmail.comwrote: I understand all that... I am just curious how many are passing the new test... is it just a few, is it a lot? J On Thu, Oct 29, 2009 at 11:20 PM, Mark Holloway m...@markholloway.com wrote: I agree with Atlanta. There are barely over 1,000 CCIE Voice folks out there. We don't want it to climb too high, too fast. I say - embrace the challenge. On Oct 29, 2009, at 6:12 PM, Aamir Panjwani wrote: Atlanta, I like your enthusiasm :) -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice- boun...@onlinestudylist.com] On Behalf Of Atlanta CCIE Sent: Friday, 30 October 2009 12:05 PM To: Jonathan Charles; Jeff Garvas; OSL Group Subject: Re: [OSL | CCIE_Voice] ccie voice stats How is it scary? I think its great! On 10/29/09, Jonathan Charles jonv...@gmail.com wrote: Well, how would we know how many people are passing? We see the occasional person saying they passed... but no other way to know. Jonathan On Thu, Oct 29, 2009 at 7:47 PM, Jeff Garvas j...@cia.net wrote: Aamir: The fewer people getting their voice ie the more you and I will be worth in the industry when we pass it. ;-) If people were passing the exam left and right I'd be more concerned that the certification is getting diluted. -Jeff On Thu, Oct 29, 2009 at 7:51 PM, Aamir Panjwani aamir.panjw...@ivision.com.au wrote: FYI all - only 5 new ccie voice in the last 32 days...scary figures J Is that because it's too challenging or just not many people attempting at the moment? http://www.networkworld.com/community/node/46893 __ 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 ___ 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 -- Sent from my mobile device CCIE# 17xxx ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ 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 ___ 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] ccie voice stats
I also should have added that I think getting your CCIE is an incredible achievement, I'm just saying spending your time achieving customers requiremnts and trust is a better way to spend your time then achieving yet another cert. Chad On Fri, Oct 30, 2009 at 12:37 AM, Jonathan Charles jonv...@gmail.comwrote: I understand all that... I am just curious how many are passing the new test... is it just a few, is it a lot? J On Thu, Oct 29, 2009 at 11:20 PM, Mark Holloway m...@markholloway.com wrote: I agree with Atlanta. There are barely over 1,000 CCIE Voice folks out there. We don't want it to climb too high, too fast. I say - embrace the challenge. On Oct 29, 2009, at 6:12 PM, Aamir Panjwani wrote: Atlanta, I like your enthusiasm :) -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice- boun...@onlinestudylist.com] On Behalf Of Atlanta CCIE Sent: Friday, 30 October 2009 12:05 PM To: Jonathan Charles; Jeff Garvas; OSL Group Subject: Re: [OSL | CCIE_Voice] ccie voice stats How is it scary? I think its great! On 10/29/09, Jonathan Charles jonv...@gmail.com wrote: Well, how would we know how many people are passing? We see the occasional person saying they passed... but no other way to know. Jonathan On Thu, Oct 29, 2009 at 7:47 PM, Jeff Garvas j...@cia.net wrote: Aamir: The fewer people getting their voice ie the more you and I will be worth in the industry when we pass it. ;-) If people were passing the exam left and right I'd be more concerned that the certification is getting diluted. -Jeff On Thu, Oct 29, 2009 at 7:51 PM, Aamir Panjwani aamir.panjw...@ivision.com.au wrote: FYI all - only 5 new ccie voice in the last 32 days...scary figures J Is that because it's too challenging or just not many people attempting at the moment? http://www.networkworld.com/community/node/46893 __ 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 ___ 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 -- Sent from my mobile device CCIE# 17xxx ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ 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 ___ 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] B-ACD just dead air...
param welcome-prompt flash:en_bacd_welcome.au it should be param welcome-prompt flash:_bacd_welcome.au because it prepends the en with paramspace english language en HTH Chad On Sat, Aug 23, 2008 at 1:26 PM, Jonathan Charles [EMAIL PROTECTED] wrote: This is an H.323 gateway, source phone is CCM, destination is CCME... same behavior when calling from an IP phone on the ccme... Jonathan On Sat, Aug 23, 2008 at 2:12 PM, Stephen Collinson [EMAIL PROTECTED] wrote: How are you calling it? PSTN or VOIP g729 or g711u? Going out on a limb here, to perhaps save a few emails. If you are calling in remotely via the GK the incoming call is perhaps g729, depending on what you set on your trunk. This voip call needs an inbound g729 voip dp to match on. When you get the dead air. Do show call active voice comp to see what the call legs are doing. Also do debug voip appl script and see what you get. HTH S -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Jonathan Charles *Sent:* 23 August 2008 19:11 *To:* OSL CCIE Voice Lab Exam *Subject:* [OSL | CCIE_Voice] B-ACD just dead air... So, I configured B-ACD (from the config on Cisco's site...) and when I call it I get dead air... ! ! interface FastEthernet0/0 ip address 10.0.0.131 255.255.255.0 speed auto no cdp log mismatch duplex h323-gateway voip interface h323-gateway voip id home ipaddr 10.0.0.63 1719 h323-gateway voip h323-id CCME h323-gateway voip tech-prefix 2# h323-gateway voip bind srcaddr 10.0.0.131 ! ! application service queue flash:app-b-acd-2.1.2.2.tcl param queue-len 15 param aa-hunt3 2001 param queue-manager-debugs 1 param aa-hunt2 2000 param number-of-hunt-grps 2 ! service aa flash:app-b-acd-aa-2.1.2.2.tcl paramspace english index 1 param number-of-hunt-grps 2 param handoff-string aa param dial-by-extension-option 1 paramspace english language en param max-time-vm-retry 2 param aa-pilot 5000 paramspace english location flash: param second-greeting-time 60 param welcome-prompt _bacd_welcome.au param call-retry-timer 15 param voice-mail 4500 param max-time-call-retry 700 param service-name queue ! global service alternate Default ! dial-peer voice 3983 voip service aa destination-pattern 5000 session target ipv4:10.0.0.131 incoming called-number 5000 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! ephone-hunt 1 sequential pilot 2000 list 3003, 3002 statistics collect ! ! ! ! ephone-hunt 2 sequential pilot 2001 list 3002, 3003 ! ! CCME#dir Directory of flash:/ 1 -rw-22201360no date c1700-spservicesk9-mz.124-15.T3.bin 2 -rw- 11650 Aug 4 2008 12:04:24 +00:00 app-cme-did-2.0.0.0.ReadMe 3 -rw- 15020 Aug 4 2008 12:04:25 +00:00 app-cme-did-2.0.0.0.tcl 5 -rw- 18836 Aug 4 2008 12:04:49 +00:00 app-b-acd-2.1.2.2-ReadMe.txt 6 -rw- 24985 Aug 4 2008 12:04:49 +00:00 app-b-acd-2.1.2.2.tcl 7 -rw- 35485 Aug 4 2008 12:04:50 +00:00 app-b-acd-aa-2.1.2.2.tcl 8 -rw- 75650 Aug 4 2008 12:04:51 +00:00 en_bacd_allagentsbusy.au 9 -rw- 83291 Aug 4 2008 12:04:52 +00:00 en_bacd_disconnect.au 10 -rw- 63055 Aug 4 2008 12:04:52 +00:00 en_bacd_enter_dest.au 11 -rw- 37952 Aug 4 2008 12:04:52 +00:00 en_bacd_invalidoption.au 12 -rw- 496521 Aug 4 2008 12:04:58 +00:00 en_bacd_music_on_hold.au 13 -rw- 123446 Aug 4 2008 12:05:00 +00:00 en_bacd_options_menu.au 14 -rw- 42978 Aug 4 2008 12:05:00 +00:00 en_bacd_welcome.au 15 -rw- 34794 Aug 4 2008 12:05:00 +00:00 en_bacd_xferto_operator.au 72 -rw- 42484no date en_dest_busy.au 73 -rw- 26376no date en_dest_unreachable.au 74 -rw- 14352no date en_disconnect.au 75 -rw- 19512no date en_enter_dest.au 76 -rw- 17167no date en_reenter_dest.au 77 -rw- 17486no date en_welcome.au 78 -rw-6627no date its-CISCO.2.0.1.0.tcl 79 -rw-3106no date its_Cisco.2.0.1.0.ReadMe Any ideas? Jonathan
Re: [OSL | CCIE_Voice] Basic Dial-Peers
of course it could but in the lab its almost guarenteed you are going to see cor (class of restriction) and therefore its best off to practice it with 911 9911... etc etc.. Chad On 7/12/08, Greg Hauser [EMAIL PROTECTED] wrote: Hello- I have a very basic h323 GW specific question. Do we need to add all these POTS dial-peers for 911; local; long distance; and international? Can not *one* PSTN dial-peer take care of all these other dial-peers? Thanks….. Example Dial-peer configs: dial-peer voice 911 pots corlist outgoing css-911-loc destination-pattern 911 port 0/0/0:0 prefix 911 ! dial-peer voice 9911 pots corlist outgoing css-911-loc destination-pattern 9911 port 0/0/0:0 prefix 911 ! dial-peer voice 7 pots corlist outgoing css-911-loc destination-pattern 9[2-9].. port 0/0/0:0 forward-digits 7 ! dial-peer voice 10 pots corlist outgoing css-911-loc destination-pattern 9[2-9]..[2-9].. port 0/0/0:0 forward-digits 10 ! dial-peer voice 11 pots corlist outgoing css-all destination-pattern 91[2-9]..[2-9].. port 0/0/0:0 forward-digits 11 ! dial-peer voice 9011 pots corlist outgoing css-all destination-pattern 9011T port 0/0/0:0 prefix 011 ! x replace with this Dial-peer: x dial-peer voice 1 pots translation-profile incoming PSTN-IN incoming called-number 3... destination-pattern 9T direct-inward-dial port 0/0/0:0 Greg J. Hauser
Re: [OSL | CCIE_Voice] Call Transfer Restriction
oh you said transfer, not sure! On 6/26/08, Chad Stachowicz [EMAIL PROTECTED] wrote: ccievoice1, you can achieve this with CFWD Calling search spaces of course! Chad On 6/26/08, ccievoice1 [EMAIL PROTECTED] wrote: Hi, Are you referring to Block OffNet to OffNet Transfer ? But that is to restrict the transferring of an external call to an external device. But IP Phones should be considered as internal device. Thanks. On Thu, Jun 26, 2008 at 8:42 PM, Cardwell, Mark [EMAIL PROTECTED] wrote: I do believe it is a system param. Transfer offnet enabled or something like that. Cheers! Mark Cardwell | System Engineer | Presidio Networked Solutions | [EMAIL PROTECTED]| Cell: 571.225.0132 | Office: 301.623.2000| FAX: 301.313.2400 -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *ccievoice1 *Sent:* Thursday, June 26, 2008 8:26 AM *To:* OSL CCIE Voice Lab Exam *Subject:* [OSL | CCIE_Voice] Call Transfer Restriction Hi, In CallManager Express, I can restrict call-transfer to only 4-digits internal DN# ! telephony-services transfer-system full-consult transfer-pattern 3... ! Just wondering, would I able to achieve the similar in CallManager? Thanks.
Re: [OSL | CCIE_Voice] CFB resources for CME
It definatly works... are you using a TAG? Chad On 6/26/08, Gregory Jost (grjost) [EMAIL PROTECTED] wrote: I can never seem to get it to register via SCCP. Greg Jost Network Consulting Engineer Unified Communications Practice Cisco Systems, Inc. 214-274-1922 -- *From:* Nguyen Le [mailto:[EMAIL PROTECTED] *Sent:* Thursday, June 26, 2008 1:24 PM *To:* Gregory Jost (grjost) *Cc:* OSL CCIE Voice Lab Exam *Subject:* Re: [OSL | CCIE_Voice] CFB resources for CME CFB will register with the HDV. However, they're never invoked in CME 3.3 On Thu, Jun 26, 2008 at 1:21 PM, Gregory Jost (grjost) [EMAIL PROTECTED] wrote: Is it even possible to register a CFB resource to CME? Are only 3-party conferences allowed? I'm guessing that the phone mixes the media for these, so a DSP resource is not required. Greg Jost Network Consulting Engineer Unified Communications Practice Cisco Systems, Inc. 214-274-1922
Re: [OSL | CCIE_Voice] POTS Dial-peer
119002001 On Jun 10, 2008, at 7:13 AM, Balamurugan Singaram [EMAIL PROTECTED] wrote: Hi, If I dial 9002001, then the outgoing number will be 900112001 or 112001 ? dial-peer voice 10 pots destination-pattern 900T forward digit all prefix 11 Thanks, Bala. dschulz [EMAIL PROTECTED] wrote: To get around this, you can set the how many digits to forward by using the forward-digits command. HTH Dave From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Balamurugan Singaram Sent: Monday, June 09, 2008 1:39 AM To: OSL CCIE Voice Lab Exam Subject: Re: [OSL | CCIE_Voice] POTS Dial-peer By default pots dial peer will strip the wildcard, so I think 2[015] will be get stripped, thanks for your reply Chand. --Bala. Chad Stachowicz [EMAIL PROTECTED] wrote: 0014152001 On Sun, Jun 8, 2008 at 9:42 PM, Balamurugan Singaram [EMAIL PROTECTED] wrote: Hi, In the following dial peer ; If I dail 2001, ougoing will be 001415201 or 0014152001, Could please let me know. dial-peer voice 31 pots destination-pattern 2[015].. port 0/0:15 prefix 0014152 Thanks, Bala. Send instant messages to your online friends http://uk.messenger.yahoo.com Send instant messages to your online friends http://uk.messenger.yahoo.com Send instant messages to your online friends http://uk.messenger.yahoo.com
Re: [OSL | CCIE_Voice] Forwarded calls from CME phones to CUE arecleared if call coming from CCM
Ah I feel great reading that other link. I ran into this studying about 5 months back, and it made me very angry. Totally a bug. Chad On 6/4/08, Vik Malhi [EMAIL PROTECTED] wrote: You can prove the transcoder is good by making a direct call into the CUE. This also would prove the allow-connections is configured correctly. If you can transcode SIP to SIP then that is big news to me! Now assuming the test above is successful see if call forwarding to CUE from another CME phone works. If not the issue is xcoder related or allow-connections being missing or indeed you might be using dial-peer 0 (which allows ANY codec). Assuming the test above works you are missing the call-forward pattern .T command. If none of the above I would need the ENTIRE configuration to expand on this answer. Vik Malhi – CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] [EMAIL PROTECTED] Join our free online support and peer group communities: http://www.IPexpert.com/communities http://www.ipexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Juan *Sent:* Thursday, May 29, 2008 4:56 AM *To:* 'OSL CCIE Voice Lab Exam' *Subject:* Re: [OSL | CCIE_Voice] Forwarded calls from CME phones to CUE arecleared if call coming from CCM Please disregard my previous mail : it seems Xcoding does indeed engage, even if the call comes from SIP g729 and it gets xcoded to g711 (direct call to CUE from CCM) In the past I think I overlooked this, as I was under the impression transcoding from SIP was not supported. Hence I thought to only have forwards to CUE work if the incoming dialpeer on CME would be h323. So, I have the same problem as you did now: no forwards to CUE work by means of the command: 'call-forward noan 3600 timeout 10' :-S When I set the DN manually to forward all to 3600, it works however... The outbound trunk on CCM is h323 (MTP checked, not waiting on h245 call capabilties, outbound fast start enabled or disabled- it doesn't matter: same as above (?) - I'd think of faststart outbound if h323-SIP...) Any help is greatly appreciated - I'm looking into it for some hours now.. I attached the ccapi output and dialpeer info from CME: BR2-RTR# May 29 2008 13:32:50.525 CEST: //209//CCAPI/cc_api_caps_ind: Call Entry Is Not Found May 29 2008 13:32:50.525 CEST: //-1/00409C510200/CCAPI/cc_api_display_ie_subfields: cc_api_call_setup_ind_common: cisco-username=2122251003 - ccCallInfo IE subfields - cisco-ani=2122251003 cisco-anitype=0 cisco-aniplan=0 cisco-anipi=0 cisco-anisi=1 dest=3001 cisco-desttype=0 cisco-destplan=0 cisco-rdie= cisco-rdn= cisco-rdntype=0 cisco-rdnplan=0 cisco-rdnpi=0 cisco-rdnsi=0 cisco-redirectreason=-1 May 29 2008 13:32:50.525 CEST: //-1/00409C510200/CCAPI/cc_api_call_setup_ind_common: Interface=0x66847600, Call Info( Calling Number=2122251003(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed), BR2-RTR#Called Number=3001(TON=Unknown, NPI=Unknown), Calling Translated=FALSE, Subsriber Type Str=Unknown, FinalDestinationFlag=TRUE, Incoming Dial-peer=2, Progress Indication=NULL(0), Calling IE Present=TRUE, Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=209 May 29 2008 13:32:50.525 CEST: //-1/00409C510200/CCAPI/ccCheckClipClir: In: Calling Number=2122251003(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed) May 29 2008 13:32:50.525 CEST: //-1/00409C510200/CCAPI/ccCheckClipClir: Out: Calling Number=2122251003(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed) May 29 2008 13:32:50.525 CEST: //209/00409C510200/CCAPI/cc_api_call_setup_ind_common: Set Up Event Sent; Call Info(Calling Number=2122251003(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed), Called Number=3001(TON=Unknown, NPI=Unknown)) May 29 2008 13:32:50.525 CEST: //209/00409C510200/CCAPI/cc_process_call_setup_ind: Event=0x66CE82F8 May 29 2008 13:32:50.525 CEST: //209/00409C510200/CCAPI/ccCallSetContext: Context=0x719A2F34 May 29 2008 13:32:50.525 CEST: //209/00409C510200/CCAPI/cc_process_call_setup_ind: CCAPI handed cid 209 with tag 2 to app _ManagedAppProcess_Default May 29 2008 13:32:50.525 CEST: //209/00409C510200/CCAPI/ccCallProceeding: Progress Indication=NULL(0) May 29 2008 13:32:50.529 CEST: //209/00409C510200/CCAPI/ccCallSetupRequest: Destination=, Calling IE Present=TRUE, Mode=0, Outgoing
Re: [OSL | CCIE_Voice] CCM NTP
Me either. : On Sat, Apr 19, 2008 at 8:02 AM, Gregory Jost (grjost) [EMAIL PROTECTED] wrote: This never works for me. 1.Stop Windows Time Service (W32Time) 2.Ensure NTP is started and startup automatic 3.Edit NTP.conf 4.Restart NTP Greg Jost Network Consulting Engineer Unified Communications Practice Cisco Systems, Inc. 214-274-1922
Re: [OSL | CCIE_Voice] B-ACD namespace error
Yeah but what that should tell you is to reload the tcl to use the new param's!!! Chad On 4/12/08, ccievoice1 [EMAIL PROTECTED] wrote: Well, you can just ignore the error message. It appeared every time you entered a param syntax. HTH On Sun, Apr 13, 2008 at 10:02 AM, Jacob Owen [EMAIL PROTECTED] wrote: Sara, I get that same error when I configure CME Scripting, but I will say that CME Scripting is one of my very weak areas so hopefully someone else will chime in and let us both know if this is an error to be concerned with or not. --- [EMAIL PROTECTED] wrote: i am testing the B-acd feature of cme when i enter the example config from lab14 on my own router, there are a few namespace error, what should i do, can anyone help? ccme-cue(config-app-param)# param handoff-string aa Warning: parameter handoff-string has not been registered under aa namespace thanks in advance Sara - GANBARE! NIPPON! Win your ticket to Olympic Games 2008. Jacob Owen CCIE #14063 (RS, Service Provider), CCVP, CCDP __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
Re: [OSL | CCIE_Voice] B-ACD namespace error
ccievoice1, What I meant to say is anytime you have B-ACD loaded and you type a change int he params, and it sasy it isn't registered under the aa namespace. You need to reload the application in order to use the new settings. CHad On 4/12/08, ccievoice1 [EMAIL PROTECTED] wrote: New params ? interesting... So what is the new param for param handoff-string param second-greeting-time param voice-mail and rest of the syntax started with param ? Please advice, thanks. On Sun, Apr 13, 2008 at 10:15 AM, Chad Stachowicz [EMAIL PROTECTED] wrote: Yeah but what that should tell you is to reload the tcl to use the new param's!!! Chad On 4/12/08, ccievoice1 [EMAIL PROTECTED] wrote: Well, you can just ignore the error message. It appeared every time you entered a param syntax. HTH On Sun, Apr 13, 2008 at 10:02 AM, Jacob Owen [EMAIL PROTECTED] wrote: Sara, I get that same error when I configure CME Scripting, but I will say that CME Scripting is one of my very weak areas so hopefully someone else will chime in and let us both know if this is an error to be concerned with or not. --- [EMAIL PROTECTED] wrote: i am testing the B-acd feature of cme when i enter the example config from lab14 on my own router, there are a few namespace error, what should i do, can anyone help? ccme-cue(config-app-param)# param handoff-string aa Warning: parameter handoff-string has not been registered under aa namespace thanks in advance Sara - GANBARE! NIPPON! Win your ticket to Olympic Games 2008. Jacob Owen CCIE #14063 (RS, Service Provider), CCVP, CCDP __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
Re: [OSL | CCIE_Voice] B-ACD namespace error
If any TCL is loaded in memory and working and you change a param it will need to be changed, however if your just putting in the params for the first time right, it should work the first time its loaded.. Cheers, Chad On 4/12/08, ccievoice1 [EMAIL PROTECTED] wrote: Oops, my bad. Sorry Chad to get you wrong. Anyway, I never needed to reload the TCL script for the param to work. I mean, in my lab it just worked after entering the necessary BACD syntax. Thanks. On Sun, Apr 13, 2008 at 10:26 AM, Chad Stachowicz [EMAIL PROTECTED] wrote: ccievoice1, What I meant to say is anytime you have B-ACD loaded and you type a change int he params, and it sasy it isn't registered under the aa namespace. You need to reload the application in order to use the new settings. CHad On 4/12/08, ccievoice1 [EMAIL PROTECTED] wrote: New params ? interesting... So what is the new param for param handoff-string param second-greeting-time param voice-mail and rest of the syntax started with param ? Please advice, thanks. On Sun, Apr 13, 2008 at 10:15 AM, Chad Stachowicz [EMAIL PROTECTED] wrote: Yeah but what that should tell you is to reload the tcl to use the new param's!!! Chad On 4/12/08, ccievoice1 [EMAIL PROTECTED] wrote: Well, you can just ignore the error message. It appeared every time you entered a param syntax. HTH On Sun, Apr 13, 2008 at 10:02 AM, Jacob Owen [EMAIL PROTECTED] wrote: Sara, I get that same error when I configure CME Scripting, but I will say that CME Scripting is one of my very weak areas so hopefully someone else will chime in and let us both know if this is an error to be concerned with or not. --- [EMAIL PROTECTED] wrote: i am testing the B-acd feature of cme when i enter the example config from lab14 on my own router, there are a few namespace error, what should i do, can anyone help? ccme-cue(config-app-param)# param handoff-string aa Warning: parameter handoff-string has not been registered under aa namespace thanks in advance Sara - GANBARE! NIPPON! Win your ticket to Olympic Games 2008. Jacob Owen CCIE #14063 (RS, Service Provider), CCVP, CCDP __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
Re: [OSL | CCIE_Voice] VOICE Passed !!!!!!!!!!!!
Yeah, the commitment that IPExpert makes to their candidates is amazing. Unmatched in the industry, and highly appreciated by all its students. Thanks guys!! Cheers, Chad On Fri, Apr 11, 2008 at 5:40 PM, Jacob Owen [EMAIL PROTECTED] wrote: Yeah, Plus the so called Practical Labs from Internetwork expert haven't even been released yet. Nothing like coming on a mailing list that is supported by IPExpert who does a fantastic job supporting voice ie candidates and talking up some other vendor that has been talking about releasing a workbook since July '07. What a T-R-O-L-L --- jason sung [EMAIL PROTECTED] wrote: Dude, A monkey can pass the test given the questions. Please keep your tips and ideas to yourself. I am sure you will pass your next CCIE using cciecert.net On Fri, Apr 11, 2008 at 7:26 PM, ccie2007 [EMAIL PROTECTED] wrote: I just passed yesterday on Tokyo I am really pleasure with this achievement First my recommendation for all guys to understand all topic of the blue print from Cisco site and documentation CD as a main resource Second I use Internetwork Expert's as practical Labs which contain a lot of the real LAB concepts, great explanation for various topics and cover almost all topics in the blue print. thanks Brain Also i really recommand that you go to cciecert.net then you will get a real ccie LAB information from this site My advice to all to go through this certificate because I have now a lot of understanding of network technology My next attempt may be CCIE SP Regards #14867 CCIE Security, R/S, VOICE Hiroyasu Kato Jacob Owen CCIE #14063 (RS, Service Provider), CCVP, CCDP __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
Re: [OSL | CCIE_Voice] Lab restrictions
all i will say is this bit me in the butt first time taking the lab, i had to have the prcotor fix my stuff about 7 minutes in :) On 4/3/08, Mark Snow [EMAIL PROTECTED] wrote: Well - there will always be for any given lab - but they also change from time to time - so posting them here wouldn't actually even help anyone! :) As Scott said - and as always with any CCIE track - READ VERY CAREFULLY. :-) Cheers, -- Mark Snow CCIE #14073 (Voice, Security) CCSI #31583 Senior Technical Instructor - IPexpert, Inc. A Cisco Learning Partner - We Accept Learning Credits! Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: [EMAIL PROTECTED] -- Join our free online support and peer group communities: http://www.IPexpert.com/communities http://www.ipexpert.com/communities -- IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- On Apr 2, 2008, at 9:51 PM, Scott Monasmith wrote: Actually, there are a few 'minor' commands you are not allowed to change/configure and they are listed on the first page of the exam. Otherwise, the lab questions will let you know whether or not you can do certain things. Read the entire exam carefully! On Wed, Apr 2, 2008 at 12:25 AM, Jonathan Charles [EMAIL PROTECTED] wrote: Actually, Jason answered my questions. On the RS lab they were very specific, from long before I took the lab that you cannot use static routes, you cannot change the ospf interface type, etc... there was a public list of commands you would not be allowed to use on the lab. There is no such list for voice, which is the answer I was looking for... Jonathan On Tue, Apr 1, 2008 at 9:31 PM, Jacob Owen [EMAIL PROTECTED] wrote: Jonathan, That is one of the big challenges of any CCIE lab, you will need to know how to do 1 thing 2-3 different ways as you won't know what you'll be allowed to do and not in the real thing. Remember, it isn't the hardest test around because it's easy, and when you pass you'll feel so much better knowing it was hard. Wish I had a better answer, but I think Jason was right on with his response. --- Jonathan Charles [EMAIL PROTECTED] wrote: Right But I am currently studying different ways of doing things... if they aren't going to let me do things, I would like to know... Jonathan On Tue, Apr 1, 2008 at 9:02 PM, jason sung [EMAIL PROTECTED] wrote: Jonathan, From my one experience, I can say that there are no restrictions universally, if there are any than it will be noted in the question. On Tue, Apr 1, 2008 at 8:24 PM, Jonathan Charles [EMAIL PROTECTED] wrote: On the RS lab you can't use static routes, etc. What restrictions are on the CCIE Voice lab? Can you not use the web interface to configure CallManager? No, seriously. Can you use ccm-manager config? That kinda stuff? Jonathan Jacob Owen CCIE #14063 (RS, Service Provider), CCVP, CCDP You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost. http://tc.deals.yahoo.com/tc/blockbuster/text5.com -- There are only 10 types of people in the world: Those who understand binary, and those who don't
Re: [OSL | CCIE_Voice] Lab restrictions
I just some thing the book said not to :) On 4/3/08, Jonathan Charles [EMAIL PROTECTED] wrote: Fix what stuff? Jonathan On Thu, Apr 3, 2008 at 5:34 PM, Chad Stachowicz [EMAIL PROTECTED] wrote: all i will say is this bit me in the butt first time taking the lab, i had to have the prcotor fix my stuff about 7 minutes in :) On 4/3/08, Mark Snow [EMAIL PROTECTED] wrote: Well - there will always be for any given lab - but they also change from time to time - so posting them here wouldn't actually even help anyone! :) As Scott said - and as always with any CCIE track - READ VERY CAREFULLY. :-) Cheers, -- Mark Snow CCIE #14073 (Voice, Security) CCSI #31583 Senior Technical Instructor - IPexpert, Inc. A Cisco Learning Partner - We Accept Learning Credits! Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: [EMAIL PROTECTED] Join our free online support and peer group communities: http://www.IPexpert.com/communities -- IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- On Apr 2, 2008, at 9:51 PM, Scott Monasmith wrote: Actually, there are a few 'minor' commands you are not allowed to change/configure and they are listed on the first page of the exam. Otherwise, the lab questions will let you know whether or not you can do certain things. Read the entire exam carefully! On Wed, Apr 2, 2008 at 12:25 AM, Jonathan Charles [EMAIL PROTECTED] wrote: Actually, Jason answered my questions. On the RS lab they were very specific, from long before I took the lab that you cannot use static routes, you cannot change the ospf interface type, etc... there was a public list of commands you would not be allowed to use on the lab. There is no such list for voice, which is the answer I was looking for... Jonathan On Tue, Apr 1, 2008 at 9:31 PM, Jacob Owen [EMAIL PROTECTED] wrote: Jonathan, That is one of the big challenges of any CCIE lab, you will need to know how to do 1 thing 2-3 different ways as you won't know what you'll be allowed to do and not in the real thing. Remember, it isn't the hardest test around because it's easy, and when you pass you'll feel so much better knowing it was hard. Wish I had a better answer, but I think Jason was right on with his response. --- Jonathan Charles [EMAIL PROTECTED] wrote: Right But I am currently studying different ways of doing things... if they aren't going to let me do things, I would like to know... Jonathan On Tue, Apr 1, 2008 at 9:02 PM, jason sung [EMAIL PROTECTED] wrote: Jonathan, From my one experience, I can say that there are no restrictions universally, if there are any than it will be noted in the question. On Tue, Apr 1, 2008 at 8:24 PM, Jonathan Charles [EMAIL PROTECTED] wrote: On the RS lab you can't use static routes, etc. What restrictions are on the CCIE Voice lab? Can you not use the web interface to configure CallManager? No, seriously. Can you use ccm-manager config? That kinda stuff? Jonathan Jacob Owen CCIE #14063 (RS, Service Provider), CCVP, CCDP You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost. http://tc.deals.yahoo.com/tc/blockbuster/text5.com -- There are only 10 types of people in the world: Those who understand binary, and those who don't
Re: [OSL | CCIE_Voice] SRST Fallback
I have noticed that my fractional PRI config comes back when the MGCP gateway registers for the first time... Is this because I have forgotten to busy out the config in CUCM? Cheers, Chad On Mon, Mar 31, 2008 at 4:14 PM, Devildoc [EMAIL PROTECTED] wrote: That's not wrong. You can include the D channel in the configure, or you don't have too. If you don't include it in the configure, then the router will include it for you. Next time, try configuring it without configuring the D channel and do a show run and you'll see the router will populate it in the configure for you. As for busying out the unused B-channels, i think Mark refers to the configuration of the T1 PRI blade for the 6500 CATOS. With CATOS, there is not a way to configure the switch to tell the CCM where to start with the B channels. It's automatically assumed to start from channel 1 if it's ascending or 23 if it's descending. So if you have a fractional T1 PRI, then you must configure the B-Channel Maintenance to indicate where the start and end of the B channels. JD Date: Mon, 31 Mar 2008 17:35:22 -0500 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] CC: ccie_voice@onlinestudylist.com; [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] SRST Fallback I have implemented a couple of fractiional PRIs, and we always include the D-channel on the controller config... Are you saying that is wrong? Also, I just listened to the audio bootcamp and you say that to do a frac PRI with MGCP, you have to busy out the other channels... Did I mishear this? There is something I am missing... Jonathan On Mon, Mar 31, 2008 at 5:26 PM, Mark Snow [EMAIL PROTECTED] wrote: That isn't actually correct. You CAN do a fractional PRI in MGCP provided that it is in IOS and do not have to busy out bchannels in UCM. And if you don't put the D-channel in, the router still knows which is the d channel based on T1 or E1. Scott, you're config looks correct -but how are you testing the failover? You can either the shut the Serial int of else write an ACL to block ports 2000 and also udp 2427 and tcp 2428. HTH, Mark Snow Sr Technical Instructor IPexpert, Inc. Sent from my iPhone On Mar 31, 2008, at 6:05 PM, Jonathan Charles [EMAIL PROTECTED] wrote: You can't do a fractional PRI in MGCP... (not unless you busy out the channels in Service Params and poll the interface on the PRI page in CCM... In other words it is a pain. Also, you forgot to include the D-Channel in your controller config... change it to: controller t1 0/0/0 pri-group timeslots 1-3,24 Jonathan On Mon, Mar 31, 2008 at 3:40 PM, Scott Monasmith [EMAIL PROTECTED] wrote: What would cause a fractional PRI to fail to come up during SRST failover? The original config is using a MGCP gateway. During the normal operation, the MGCP gateway works just fine. However, during SRST, the PRI never comes up. I keep seeing TEI_ASSIGNED. isdn switch-type primary-ni controller t1 0/0/0 pri-group timeslots 1-3 service mgcp int serial 0/0/0:23 isdn bind-l3 ccm-manager mgcp call-agent 10.1.200.22 mgcp ccm-manager mgcp ccm-manager fallback-mgcp ccm-manager music-on-hold ccm-manager switch immedate ccm-manager redundant 10.1.200.21 application global service alternate default call-manager-fallback max-ephone 10 max-dn 20 source ip-address 172.2.100.1 -- Test your Star IQ Play now!http://club.live.com/red_carpet_reveal.aspx?icid=redcarpet_HMTAGMAR
Re: [OSL | CCIE_Voice] Trailing #
Indeed it is assumed in DCM to drop a trailing # sign On Mar 14, 2008, at 9:27 PM, Jonathan Charles [EMAIL PROTECTED] wrote: I just read that the trailing-# DDI only applies to the @ pattern... is this true? Jonathan
Re: [OSL | CCIE_Voice] Trailing #
Ccm On Mar 15, 2008, at 9:42 PM, Chad Stachowicz [EMAIL PROTECTED] wrote: Indeed it is assumed in DCM to drop a trailing # sign On Mar 14, 2008, at 9:27 PM, Jonathan Charles [EMAIL PROTECTED] wrote: I just read that the trailing-# DDI only applies to the @ pattern... is this true? Jonathan
Re: [OSL | CCIE_Voice] IPCC Phone Agent Service URL - moved from Univercd ??
Mike, I can tell you at San Jose this is not the case, my off site redirect links worked Chad On 2/29/08, Mike Prestidge [EMAIL PROTECTED] wrote: Hi Scott, it sounds like there are some discrepancies then depending on where you sit the lab? I had my first attempt two weeks ago and this was one of the things I tested during the lab. I tried to reach multiple documents that had been moved off the univercd site. In my case even waiting for the automatic redirect failed. Mike From: Scott Monasmith [mailto:[EMAIL PROTECTED] Sent: Sat 1/03/2008 10:34 a.m. To: Mike Prestidge Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IPCC Phone Agent Service URL - moved from Univercd ?? This is a common misconception, my friend. It does redirect you outside of UniverCD, however, on the redirect page it will prompt you to either click on the redirected link or wait 10 seconds for your browser to do it automatically. If you wait 10 seconds, it will redirect you just fine. However, if you click on the link instead it will fail. Trust me. Cheers, Scott On Fri, Feb 29, 2008 at 3:25 PM, Mike Prestidge [EMAIL PROTECTED] wrote: Hi Scott, the path below is what I had originally used to find the IP Phone Agent URL, however the fact that it now redirects to a link outside of cisco.com/univercd means that it is not reachable during the lab exam. Mike From: Scott Monasmith [mailto:[EMAIL PROTECTED] Sent: Sat 1/03/2008 3:56 a.m. To: Mike Prestidge Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IPCC Phone Agent Service URL - moved from Univercd ?? Here is the path for IP Phone agent URL - DocCD - Customer Contact Center - Cisco IPCC Express and IP IVR - Cisco Customer Response Solution 5.0(x) - English - Documentation for Cisco IP Agents - Cisco CAD Installation Guide CAD 6.4 for Unified -- This will redirect you to a .pdf file. Do a search for http:// inside this .PDF and you'll eventually find the URL. However, this document does not tell you what variables you need to setup (Pwd, Ext, ID) for the IP Phone service. here is the direct link. http://www.cisco.com/univercd/cc/td/doc/product/voice/sw_ap_to/apps_5_0/english/agents/cad64ig.pdf Cheers, Scott On Tue, Feb 26, 2008 at 4:22 PM, Mike Prestidge [EMAIL PROTECTED] wrote: It seems that the (annoying) people moving the documents off Univercd have now also moved the documents with the URL for IP Phone agents!! I used to be able to find this by browsing via the following: Univercd Customer Contact Software IPCC Express and IP IVR CRS 5.0(x) English Documentation for Cisco IP Agents Cisco CAD Installation Guide 6.4 Now this documentation has also been moved to a link that is not available in the lab. Does anyone know an alternative location to find the URL within Univercd? Mike This communication, including any attachments, is confidential. If you are not the intended recipient, you should not read it - please contact me immediately, destroy it, and do not copy or use any part of this communication or disclose anything about it. Thank you. Please note that this communication does not designate an information system for the purposes of the Electronic Transactions Act 2002. -- There are only 10 types of people in the world: Those who understand binary, and those who don't -- There are only 10 types of people in the world: Those who understand binary, and those who don't
[OSL | CCIE_Voice] CME Display question
you know on call manager how the external phone number mask will populate the top right portion of a phone display, in the black bar at the top. Is there a particular field that will change this? I have figured out a dodgy way using secondary numbers on my ephone dn, but I wondered if there was another way to make it stray from the default of using button 1's primary number.. Thanks, Chad
[OSL | CCIE_Voice] sip fxs
in the ipexpert book it jsut shows registering the gateway to sip. How do i specify toi only register the FXP port? Thanks Chad
[OSL | CCIE_Voice] Locations
Guys I know this is silly but its the last day before my lab, and I'm having somme jitters G711 in location 64 or 80kbps? Thanks Chad
Re: [OSL | CCIE_Voice] Multicast MOH
Mark, so your saying just IP pim dense mode no ip pim sparse-dense-mode? Thanks, Chad On Feb 13, 2008 9:44 AM, Mark Snow [EMAIL PROTECTED] wrote: Each router only decrements 1 hop per router - not interface - so technically it would be 2 hops away. Every interface involved in Multicast needs to have PIM enabled in some sort of fashion (including the Loopback interface if streaming out to a PSTN POTS Trunk!) - the easiest for the Voice lab being PIM- Dense where we flood everywhere and do not care about RPs. Bear in mind you will not hear Multicast if you phone HQ Phone 3 to BR1 Phone 3 (where both those phones are IPBlue or even hardware phones behind your EasyVPN connection) and hit hold - due to the fact that IPSec only supports Unicast traffic - not Multicast or Broadcast traffic. You CAN hear it however by picking up lets say BR1 Phone 3 and phoning out to the PSTN (911 for ease) and putting the PSTN Phone on hold by pressing hold from the BR1 Phone. This puts the BR1 Gateway on hold technically - and the BR1 Gateway turns the VoIP - into regular old PCM - sends it out the PRI - over to the PSTN Gateway PRI - then becomes VoIP again (this time Unicast) and streams to the PSTN Phone. Assuming you are controlling your PSTN Phone with an IPBlue client (we show you how on the ProctorLabs Voice vRack web page) - then you should hear the stream and be able to do a sh ip mroute along with the PerfMon tool on the UCM Server - and verify that Multicast is indeed working and you can hear it! Cheers, Mark Snow CCIE #14073 (Voice, Security) CCSI #31583 Senior Technical Instructor - IPexpert, Inc. A Cisco Learning Partner - We Accept Learning Credits! Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: [EMAIL PROTECTED] IPexpert - The Global Leader in Self-Study, Classroom-Based, Video On Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. On Feb 13, 2008, at 12:06 AM, jason sung wrote: I am little confused on Multicast MOH hops. Can somebody correct me if I am wrong about the following. 1. MOH server is in the same vlan as voice vlan In this case if I were to stream MMOH to Branch 1: My hop count will be 3. 1st hop is local voice vlan, 2nd hop is the WAN link and 3rd hop is the branch 1 phones. 2. MOH server is in a seperate server vlan. In this case if I were to stream MMOH to Branch 1: My hop count will still be 3. 1st hop is server vlan, 2nd hop is my WAN link and 3rd hop is the branch 1 phones. Do I need to place ip pim-dense mode command on the serial interfaces as well? Jason.
Re: [OSL | CCIE_Voice] Can't get Virtual-Template to work
I also have a question along the same lines. Can we do this with standard frame relay traffic shapping and not VATS? is VATS a requirement for ppp? Thanks, Chad On 2/13/08, Devildoc [EMAIL PROTECTED] wrote: Can someone please let me know what I am missing from my configuration below for the Virtual-Template configuration? I can't seem to get the interface working. It's in the down state. Any help is greatly appreciated. Thanks. Here is my configuration --- class-map match-any SIGNAL match ip dscp cs3 af31 class-map match-all VOICE match ip dscp ef policy-map LLQ class VOICE priority percent 33 compress header ip rtp class SIGNAL bandwidth percent 5 class class-default fair-queue random-detect policy-map CB-VATS class class-default shape average 729600 3648 0 shape adaptive 364800 shape fr-voice-adapt deactivation 30 service-policy LLQ interface Serial0/1/0:0 no ip address encapsulation frame-relay IETF frame-relay fragmentation voice-adaptive deactivation 30 frame-relay lmi-type ansi interface Serial0/1/0:0.1 point-to-point ip address 162.1.101.1 255.255.255.0 ip ospf mtu-ignore frame-relay interface-dlci 201 class WAN-EDGE interface Serial0/1/0:0.2 point-to-point ip ospf mtu-ignore frame-relay interface-dlci 202 ppp Virtual-Template1 interface Virtual-Template1 bandwidth 768 ip address 162.1.102.1 255.255.255.0 ip ospf mtu-ignore ppp multilink ppp multilink fragment delay 10 ppp multilink interleave service-policy output CB-VATS map-class frame-relay WAN-EDGE frame-relay fragment 480 service-policy output CB-VATS -- Helping your favorite cause is as easy as instant messaging. You IM, we give. Learn more.http://im.live.com/Messenger/IM/Home/?source=text_hotmail_join
Re: [OSL | CCIE_Voice] Can't get Virtual-Template to work
I'm sorry, Can someone please verify I did the correct method of nesting policy maps assuming TOHQLLQ was an LLQ policy-map TOHQLLQ policy-map TOHQ class class-default shape average 729600 shape adaptive 729600 shape fr-voice-adapt deactivation 30 service-policy TOHQLLQ Also devil remember anything that is configured to fragment always needs to be matched on the other side of the PVC, frame relay traffic shapping on the other end although it should be done, can happen one way without being service impacting. If you have fragmentation correctly configured on both ends of the PVC you should see the OSPF adjencies come up :) thats my trick On 2/13/08, anil batra [EMAIL PROTECTED] wrote: You are applying LFI on virtual template as well as on map-class which is also not correct. -Anil --- Devildoc [EMAIL PROTECTED] wrote: Can someone please let me know what I am missing from my configuration below for the Virtual-Template configuration? I can't seem to get the interface working. It's in the down state. Any help is greatly appreciated. Thanks. Here is my configuration --- class-map match-any SIGNAL match ip dscp cs3 af31class-map match-all VOICE match ip dscp ef policy-map LLQ class VOICE priority percent 33 compress header ip rtp class SIGNAL bandwidth percent 5 class class-default fair-queue random-detect policy-map CB-VATS class class-default shape average 729600 3648 0 shape adaptive 364800 shape fr-voice-adapt deactivation 30 service-policy LLQ interface Serial0/1/0:0 no ip address encapsulation frame-relay IETF frame-relay fragmentation voice-adaptive deactivation 30 frame-relay lmi-type ansiinterface Serial0/1/0:0.1 point-to-point ip address 162.1.101.1 255.255.255.0 ip ospf mtu-ignore frame-relay interface-dlci 201 class WAN-EDGEinterface Serial0/1/0:0.2 point-to-point ip ospf mtu-ignore frame-relay interface-dlci 202 ppp Virtual-Template1interface Virtual-Template1 bandwidth 768 ip address 162.1.102.1 255.255.255.0 ip ospf mtu-ignore ppp multilink ppp multilink fragment delay 10 ppp multilink interleave service-policy output CB-VATSmap-class frame-relay WAN-EDGE frame-relay fragment 480 service-policy output CB-VATS _ Helping your favorite cause is as easy as instant messaging. You IM, we give. http://im.live.com/Messenger/IM/Home/?source=text_hotmail_join Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ
[OSL | CCIE_Voice] CME TCL B-ACD
below is the relevant config, I don't understand why no outgoing dial-peer is being matched? Any help? Thanks ! application service queue flash:app-b-acd-2.1.0.0.tcl param queue-len 20 param number-of-hunt-grps 1 param aa-hunt2 3210 ! service aa flash:app-b-acd-aa-2.1.0.0.tcl paramspace english index 1 param number-of-hunt-grps 1 param dial-by-extension-option 4 param handoff-string aa paramspace english language en param max-time-vm-retry 2 param max-extension-length 4 param aa-pilot 3200 paramspace english location flash: param second-greeting-time 30 param welcome-prompt en_bacd_welcome.au param call-retry-timer 15 param max-time-call-retry 600 param voice-mail 3600 param service-name queue ! ! ! ! controller E1 0/0/0 ! ! ! ! ! interface Loopback0 ip address 172.4.102.1 255.255.255.255 ip ospf network point-to-point h323-gateway voip interface h323-gateway voip bind srcaddr 172.4.102.1 ! interface FastEthernet0/0 no ip address duplex auto speed auto ! interface FastEthernet0/0.140 encapsulation dot1Q 140 native no snmp trap link-status ! interface FastEthernet0/0.240 encapsulation dot1Q 240 ip address 10.4.202.1 255.255.255.0 no snmp trap link-status ! interface Service-Engine0/0 no ip address shutdown ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! interface Serial0/1/0 no ip address encapsulation frame-relay IETF no fair-queue frame-relay lmi-type ansi ! interface Serial0/1/0.1 point-to-point ip address 162.4.102.2 255.255.255.0 frame-relay interface-dlci 102 ! router ospf 1 log-adjacency-changes network 10.4.102.0 0.0.0.255 area 0 network 10.4.202.0 0.0.0.255 area 0 network 162.4.102.0 0.0.0.255 area 0 network 172.4.102.0 0.0.0.255 area 0 ! ip classless ! ! ip http server no ip http secure-server ! ! ! ! tftp-server flash:P00307010100.bin tftp-server flash:P00303020214.bin tftp-server flash:P00403020214.bin tftp-server flash:P00305000600.sbn tftp-server flash:P00307020200.bin tftp-server flash:P00307020200.loads tftp-server flash:P00307020200.sb2 tftp-server flash:P00307020200.sbn ! control-plane ! ! ! ! ! ! ! dial-peer voice 10 voip service aa destination-pattern 3200 ! dial-peer voice 20 voip session target ipv4:10.4.202.1 incoming called-number . ! dial-peer voice 35 voip service aa destination-pattern 3313243200 ! ! ! ! gatekeeper shutdown ! ! telephony-service load 7910 P00403020214 load 7960-7940 P00307020200 max-ephones 15 max-dn 20 ip source-address 10.4.202.1 port 2000 create cnf-files version-stamp Jan 01 2002 00:00:00 dialplan-pattern 1 3313243... extension-length 4 max-conferences 8 gain -6 call-forward pattern .T moh moh_file.wav transfer-system full-consult transfer-pattern .T ! ! ephone-dn 1 dual-line number 3001 caller-id block ! ! ephone-dn 2 dual-line number 3002 ! ! ephone-dn 9 number 3999 mwi on ! ! ephone-dn 10 number 3998 mwi off ! ! ephone 1 mac-address 0030.94C4.22D6 ! ! ! ephone 2 mac-address 0011.BBE1.ADDF ! ! ephone-hunt 1 peer pilot 3210 list 3001, 3002 timeout 12 statistics collect ! ! ! line con 0 line aux 0 line 194 no activation-character no exec transport preferred none transport input all transport output all line vty 0 4 privilege level 15 transport input telnet line vty 5 15 privilege level 15 transport input telnet ! warm-reboot scheduler allocate 2 1000 ! end P4-BR2-RTR(config-dial-peer)# *Feb 12 23:55:57.339: //-1/00AE360E0800/DPM/dpAssociateIncomingPeerCore: Calling Number=1000, Called Number=3200, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH *Feb 12 23:55:57.339: //-1/00AE360E0800/DPM/dpAssociateIncomingPeerCore: Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=20 *Feb 12 23:55:57.339: //-1/00AE360E0800/DPM/dpAssociateIncomingPeerCore: Calling Number=1000, Called Number=3200, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH *Feb 12 23:55:57.343: //-1/00AE360E0800/DPM/dpAssociateIncomingPeerCore: Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=20 *Feb 12 23:55:57.347: //-1/00AE360E0800/DPM/dpMatchPeersCore: Calling Number=, Called Number=3200, Peer Info Type=DIALPEER_INFO_SPEECH *Feb 12 23:55:57.347: //-1/00AE360E0800/DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=3200 *Feb 12 23:55:57.347: //-1/00AE360E0800/DPM/dpMatchPeersCore: No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1) *Feb 12 23:55:57.347: //-1/00
Re: [OSL | CCIE_Voice] CME TCL B-ACD
patel, just have 2 incoming dial peers ;) Chad On 2/12/08, Patel, Mrugesh [EMAIL PROTECTED] wrote: So, we have to hair-pin the call back in from PSTN or IPIPGW. No? *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Allam Hassan *Sent:* Tuesday, February 12, 2008 6:18 PM *To:* Chad Stachowicz *Cc:* CCIE Maillist *Subject:* Re: [OSL | CCIE_Voice] CME TCL B-ACD Service Applications cannot be triggered on Outbound Dial Peers ... only Inbound Dial Peers. On Feb 12, 2008, at 6:59 PM, Chad Stachowicz wrote: below is the relevant config, I don't understand why no outgoing dial-peer is being matched? Any help? Thanks ! application service queue flash:app-b-acd-2.1.0.0.tcl param queue-len 20 param number-of-hunt-grps 1 param aa-hunt2 3210 ! service aa flash:app-b-acd-aa-2.1.0.0.tcl paramspace english index 1 param number-of-hunt-grps 1 param dial-by-extension-option 4 param handoff-string aa paramspace english language en param max-time-vm-retry 2 param max-extension-length 4 param aa-pilot 3200 paramspace english location flash: param second-greeting-time 30 param welcome-prompt en_bacd_welcome.au param call-retry-timer 15 param max-time-call-retry 600 param voice-mail 3600 param service-name queue ! ! ! ! controller E1 0/0/0 ! ! ! ! ! interface Loopback0 ip address 172.4.102.1 255.255.255.255 ip ospf network point-to-point h323-gateway voip interface h323-gateway voip bind srcaddr 172.4.102.1 ! interface FastEthernet0/0 no ip address duplex auto speed auto ! interface FastEthernet0/0.140 encapsulation dot1Q 140 native no snmp trap link-status ! interface FastEthernet0/0.240 encapsulation dot1Q 240 ip address 10.4.202.1 255.255.255.0 no snmp trap link-status ! interface Service-Engine0/0 no ip address shutdown ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! interface Serial0/1/0 no ip address encapsulation frame-relay IETF no fair-queue frame-relay lmi-type ansi ! interface Serial0/1/0.1 point-to-point ip address 162.4.102.2 255.255.255.0 frame-relay interface-dlci 102 ! router ospf 1 log-adjacency-changes network 10.4.102.0 0.0.0.255 area 0 network 10.4.202.0 0.0.0.255 area 0 network 162.4.102.0 0.0.0.255 area 0 network 172.4.102.0 0.0.0.255 area 0 ! ip classless ! ! ip http server no ip http secure-server ! ! ! ! tftp-server flash:P00307010100.bin tftp-server flash:P00303020214.bin tftp-server flash:P00403020214.bin tftp-server flash:P00305000600.sbn tftp-server flash:P00307020200.bin tftp-server flash:P00307020200.loads tftp-server flash:P00307020200.sb2 tftp-server flash:P00307020200.sbn ! control-plane ! ! ! ! ! ! ! dial-peer voice 10 voip service aa destination-pattern 3200 ! dial-peer voice 20 voip session target ipv4:10.4.202.1 incoming called-number . ! dial-peer voice 35 voip service aa destination-pattern 3313243200 ! ! ! ! gatekeeper shutdown ! ! telephony-service load 7910 P00403020214 load 7960-7940 P00307020200 max-ephones 15 max-dn 20 ip source-address 10.4.202.1 port 2000 create cnf-files version-stamp Jan 01 2002 00:00:00 dialplan-pattern 1 3313243... extension-length 4 max-conferences 8 gain -6 call-forward pattern .T moh moh_file.wav transfer-system full-consult transfer-pattern .T ! ! ephone-dn 1 dual-line number 3001 caller-id block ! ! ephone-dn 2 dual-line number 3002 ! ! ephone-dn 9 number 3999 mwi on ! ! ephone-dn 10 number 3998 mwi off ! ! ephone 1 mac-address 0030.94C4.22D6 ! ! ! ephone 2 mac-address 0011.BBE1.ADDF ! ! ephone-hunt 1 peer pilot 3210 list 3001, 3002 timeout 12 statistics collect ! ! ! line con 0 line aux 0 line 194 no activation-character no exec transport preferred none transport input all transport output all line vty 0 4 privilege level 15 transport input telnet line vty 5 15 privilege level 15 transport input telnet ! warm-reboot scheduler allocate 2 1000 ! end P4-BR2-RTR(config-dial-peer)# *Feb 12 23:55:57.339: //-1/00AE360E0800/DPM/dpAssociateIncomingPeerCore: Calling Number=1000, Called Number=3200, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH *Feb 12 23:55:57.339: //-1/00AE360E0800/DPM/dpAssociateIncomingPeerCore: Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=20 *Feb 12 23:55:57.339: //-1/00AE360E0800/DPM/dpAssociateIncomingPeerCore: Calling Number=1000, Called Number=3200, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH *Feb 12 23:55:57.343: //-1/00AE360E0800/DPM/dpAssociateIncomingPeerCore: Result=Success(0) after
Re: [OSL | CCIE_Voice] CME TCL B-ACD
Alright I ahve it connecting, buts its not playing any audio or respdonding to key presses. Anything come to the top of anyone's head? Thanks Chad On 2/12/08, Patel, Mrugesh [EMAIL PROTECTED] wrote: I always think far. J *From:* Chad Stachowicz [mailto:[EMAIL PROTECTED] *Sent:* Tuesday, February 12, 2008 7:15 PM *To:* Patel, Mrugesh *Cc:* Allam Hassan; CCIE Maillist *Subject:* Re: [OSL | CCIE_Voice] CME TCL B-ACD patel, just have 2 incoming dial peers ;) Chad On 2/12/08, *Patel, Mrugesh* [EMAIL PROTECTED] wrote: So, we have to hair-pin the call back in from PSTN or IPIPGW. No? *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Allam Hassan *Sent:* Tuesday, February 12, 2008 6:18 PM *To:* Chad Stachowicz *Cc:* CCIE Maillist *Subject:* Re: [OSL | CCIE_Voice] CME TCL B-ACD Service Applications cannot be triggered on Outbound Dial Peers ... only Inbound Dial Peers. On Feb 12, 2008, at 6:59 PM, Chad Stachowicz wrote: below is the relevant config, I don't understand why no outgoing dial-peer is being matched? Any help? Thanks ! application service queue flash:app-b-acd-2.1.0.0.tcl param queue-len 20 param number-of-hunt-grps 1 param aa-hunt2 3210 ! service aa flash:app-b-acd-aa-2.1.0.0.tcl paramspace english index 1 param number-of-hunt-grps 1 param dial-by-extension-option 4 param handoff-string aa paramspace english language en param max-time-vm-retry 2 param max-extension-length 4 param aa-pilot 3200 paramspace english location flash: param second-greeting-time 30 param welcome-prompt en_bacd_welcome.au param call-retry-timer 15 param max-time-call-retry 600 param voice-mail 3600 param service-name queue ! ! ! ! controller E1 0/0/0 ! ! ! ! ! interface Loopback0 ip address 172.4.102.1 255.255.255.255 ip ospf network point-to-point h323-gateway voip interface h323-gateway voip bind srcaddr 172.4.102.1 ! interface FastEthernet0/0 no ip address duplex auto speed auto ! interface FastEthernet0/0.140 encapsulation dot1Q 140 native no snmp trap link-status ! interface FastEthernet0/0.240 encapsulation dot1Q 240 ip address 10.4.202.1 255.255.255.0 no snmp trap link-status ! interface Service-Engine0/0 no ip address shutdown ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! interface Serial0/1/0 no ip address encapsulation frame-relay IETF no fair-queue frame-relay lmi-type ansi ! interface Serial0/1/0.1 point-to-point ip address 162.4.102.2 255.255.255.0 frame-relay interface-dlci 102 ! router ospf 1 log-adjacency-changes network 10.4.102.0 0.0.0.255 area 0 network 10.4.202.0 0.0.0.255 area 0 network 162.4.102.0 0.0.0.255 area 0 network 172.4.102.0 0.0.0.255 area 0 ! ip classless ! ! ip http server no ip http secure-server ! ! ! ! tftp-server flash:P00307010100.bin tftp-server flash:P00303020214.bin tftp-server flash:P00403020214.bin tftp-server flash:P00305000600.sbn tftp-server flash:P00307020200.bin tftp-server flash:P00307020200.loads tftp-server flash:P00307020200.sb2 tftp-server flash:P00307020200.sbn ! control-plane ! ! ! ! ! ! ! dial-peer voice 10 voip service aa destination-pattern 3200 ! dial-peer voice 20 voip session target ipv4:10.4.202.1 incoming called-number . ! dial-peer voice 35 voip service aa destination-pattern 3313243200 ! ! ! ! gatekeeper shutdown ! ! telephony-service load 7910 P00403020214 load 7960-7940 P00307020200 max-ephones 15 max-dn 20 ip source-address 10.4.202.1 port 2000 create cnf-files version-stamp Jan 01 2002 00:00:00 dialplan-pattern 1 3313243... extension-length 4 max-conferences 8 gain -6 call-forward pattern .T moh moh_file.wav transfer-system full-consult transfer-pattern .T ! ! ephone-dn 1 dual-line number 3001 caller-id block ! ! ephone-dn 2 dual-line number 3002 ! ! ephone-dn 9 number 3999 mwi on ! ! ephone-dn 10 number 3998 mwi off ! ! ephone 1 mac-address 0030.94C4.22D6 ! ! ! ephone 2 mac-address 0011.BBE1.ADDF ! ! ephone-hunt 1 peer pilot 3210 list 3001, 3002 timeout 12 statistics collect ! ! ! line con 0 line aux 0 line 194 no activation-character no exec transport preferred none transport input all transport output all line vty 0 4 privilege level 15 transport input telnet line vty 5 15 privilege level 15 transport input telnet ! warm-reboot scheduler allocate 2 1000 ! end P4-BR2-RTR(config-dial-peer)# *Feb 12 23:55:57.339: //-1/00AE360E0800/DPM/dpAssociateIncomingPeerCore: Calling Number=1000, Called Number=3200, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH *Feb 12 23:55:57.339: //-1/00AE360E0800/DPM/dpAssociateIncomingPeerCore
[OSL | CCIE_Voice] CCM-H323g729-IPIPGW-sipg711-cue FAIL
any idea why we can't do this? It doesn't even try a call forward if I bring it in as sipg711ulaw? dtmf problem? but even if it was dtmf I figure they would still try and transfer in Thanks, Chad
[OSL | CCIE_Voice] Multicast MOH
guys, can we get mutlitcast MOH to out ipcommunicator? I see multicast incrementing on interfaces to BR1 with sh ip multicast interface int but can't hear it Thanks, Chad
[OSL | CCIE_Voice] DTMF-Relay once over
All, I'm looking for a good read or guide for understanding dtmf-relay. Mainly I'm trying to understand what to modify and why I do it for H323 to Sip conversions. Thanks, Chad
[OSL | CCIE_Voice] CME tftp-server commands
Guys, I have never had to put any tftp server commands or define any loads for any phone types under telephony service. Is there any reason to do this unless we are asked to upgrade the firmwares in the lab? Thanks, Chad
[OSL | CCIE_Voice] QOS Confirmation
all, sadly in the last week they have migrated all of the Call manager docs off of univercd and including the nice ports doc. from a prior post from vik Couple of issues. For a full list of port numbers see link below. I've modifiedthe ACL with the SIP port #'s and removed the unused H323 port numbers. Also each line within the ACL will need the policer applied if you are policing all signaling traffic. http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/sec_vir/udp_tcp/41plrev2.pdf set qos policed-dscp-map 24,26:10 set qos policer aggregate POLICE-CCM rate 32 burst 8000 policed-dscp set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM tcp any range 2000 2002 any set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM tcp any eq 2428 any set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM udp any eq 2427 any set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM tcp any any eq 1720 set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM tcp any any eq 1718 set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM udp any eq 1719 any set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM tcp any any eq 5060 set qos acl ip CCM-SIGNAL dscp 24 aggregate POLICE-CCM udp any any eq 5060 commit qos acl CCM-SIGNAL set qos acl map CCM-SIGNAL 3/3 I was wondering i think that 1719 is actually a destination port not a source port and it is wrong in this? Can anyone confirm or deny? Thanks, Chad
Re: [OSL | CCIE_Voice] stripping 1# at hq site and CME site
Patel, PSTN wouldn't send that now would it? the only way you going to be recieving a tech prefix of #anything is over VOIP Chad On 2/8/08, Patel, Mrugesh [EMAIL PROTECTED] wrote: I meant if the call to CME was being handed off from PSTN, and HQ site would be dialing *#2*5552223001 Dial-peer voice 100 pots Incoming-called number #2………. *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Patel, Mrugesh *Sent:* Friday, February 08, 2008 9:20 AM *To:* Strong Frog; Mark Snow *Cc:* CCIE Voice Maillist; Toltzien, Matt; Devildoc *Subject:* Re: [OSL | CCIE_Voice] stripping 1# at hq site and CME site Instead of voice translation rules or num-exp, can we also use the following? IF CME has 3xxx numbers and #2 is the tech-prefix. Dial-peer voice 100 pots Incoming-called number #2…. Since this is a pots dial-peer, it should strip of the matching #2. No? *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Strong Frog *Sent:* Friday, February 08, 2008 8:45 AM *To:* Mark Snow *Cc:* CCIE Voice Maillist; Toltzien, Matt; Devildoc *Subject:* Re: [OSL | CCIE_Voice] stripping 1# at hq site and CME site Thanks to all who replied to this post... Totally agreed with mark Matt. Coz the digit could vary so we've to use xlation pattern in CCM and CME site. No other way around? Another question: CME site ext: CCM site ext: Call flow: === 1. CME phone - calls HQ phone , via GK. 2. CME VoIP ras dial-peer prefixes 1# before sending call to GK. 3. GK receives 1# 4. CCM receives 1# 5. Xlation pattern in CCM strips 1# and phone finally rings... In what circumstances CCM may receive 1 digits from CME site instead of 1# ? I couldn't recall this question completely but one of my friend tells me that be careful that CCM may receive this kind of digits so my xlation pattern on cme (striping 1# may not work). Frog
Re: [OSL | CCIE_Voice] Gateway Location
Vik, I had a problem yesterday when a call going out my pstn at HQ into my BR1 router MGCP would fail if BR1 location was set... also changing the difference of 64 vs 128 fixed the problem in location. however this is just call signaling in this case since it was over the PSTN Any input? Chad On 1/31/08, Vik Malhi [EMAIL PROTECTED] wrote: Yes you should have Location set on the gateway. Take the scenario whereby a BR1 phone calls out of the BR1 gateway (e.g. 911, local calls). If the gateway is not configured to be in the same location as the phone, then from CallManager's perspective this call is a call over the WAN and hence a deducation in location bandwidth will occur. Even though the gateway and phone are on the same LAN! You MUST put the remote gateway in the same location as the remote phone to prevent CAC from kicking in on calls between devices on the same LAN. When you set the location on an IOS MGCP gateway, unless you used the ccm-manager config commands, you must reset the gateway in IOS (as opposed to the Reset button on the web page). To do this type no mgcp, Carriage Return mgcp. Vik Malhi CCIE Voice Instructor / Developer - IPexpert, Inc. CCIE Voice #13890 CCSI #31584 URL: http://www.IPexpert.com http://www.ipexpert.com/ Toll Free: +1.866.225.8064 International: +1.810.326.1444 -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Chad Stachowicz *Sent:* Wednesday, January 30, 2008 8:33 PM *To:* CCIE Maillist *Subject:* [OSL | CCIE_Voice] Gateway Location Should we ever set a location on the gateway? I figured we should, however it seems to break most of my labs :) Thanks, Chad
Re: [OSL | CCIE_Voice] Gateway Location + Trunk Location
I believe its a bad Idea to use locations with gatekeeper CAC gatekeeper CAC + Locations CAC = bad.. GK should be location unlimited or set to NONE IMO On 1/31/08, Balamurugan Singaram [EMAIL PROTECTED] wrote: Hi, In CCM, Trunk it is fine to apply the location or just apply none in Trunk and apply GK bandwidth in IOS command. Could you please suggest me? Thanks, Bala. *Vik Malhi [EMAIL PROTECTED]* wrote: because 64 failed and 128 worked, it means the negotiated codec was g711 (80kbps of location bandwidth) and the BR1 phone and the gatway are in different locations. You MUST no mgcp/mgcp after setting the location and ensure the BR1 phone is in the same location as the gateway. Vik Malhi CCIE Voice Instructor / Developer - IPexpert, Inc. CCIE Voice #13890 CCSI #31584 URL: http://www.IPexpert.com http://www.ipexpert.com/ Toll Free: +1.866.225.8064 International: +1.810.326.1444 -- *From:* Chad Stachowicz [mailto:[EMAIL PROTECTED] *Sent:* Thursday, January 31, 2008 3:41 PM *To:* [EMAIL PROTECTED] *Cc:* CCIE Maillist *Subject:* Re: [OSL | CCIE_Voice] Gateway Location Vik, I had a problem yesterday when a call going out my pstn at HQ into my BR1 router MGCP would fail if BR1 location was set... also changing the difference of 64 vs 128 fixed the problem in location. however this is just call signaling in this case since it was over the PSTN Any input? Chad On 1/31/08, Vik Malhi [EMAIL PROTECTED] wrote: Yes you should have Location set on the gateway. Take the scenario whereby a BR1 phone calls out of the BR1 gateway (e.g. 911, local calls). If the gateway is not configured to be in the same location as the phone, then from CallManager's perspective this call is a call over the WAN and hence a deducation in location bandwidth will occur. Even though the gateway and phone are on the same LAN! You MUST put the remote gateway in the same location as the remote phone to prevent CAC from kicking in on calls between devices on the same LAN. When you set the location on an IOS MGCP gateway, unless you used the ccm-manager config commands, you must reset the gateway in IOS (as opposed to the Reset button on the web page). To do this type no mgcp, Carriage Return mgcp. Vik Malhi CCIE Voice Instructor / Developer - IPexpert, Inc. CCIE Voice #13890 CCSI #31584 URL: http://www.IPexpert.com http://www.ipexpert.com/ Toll Free: +1.866.225.8064 International: +1.810.326.1444 -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Chad Stachowicz *Sent:* Wednesday, January 30, 2008 8:33 PM *To:* CCIE Maillist *Subject:* [OSL | CCIE_Voice] Gateway Location Should we ever set a location on the gateway? I figured we should, however it seems to break most of my labs :) Thanks, Chad Send instant messages to your online friends http://uk.messenger.yahoo.com
Re: [OSL | CCIE_Voice] IPCC Question
awesome thanks a bunch guys! On 1/30/08, Devildoc [EMAIL PROTECTED] wrote: That's right! The first name, last name, username and password should be set to telecaster. JD -- Date: Wed, 30 Jan 2008 11:28:30 +0800 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IPCC Question I think password should be *telecaster* On Jan 30, 2008 11:24 AM, Chad Stachowicz [EMAIL PROTECTED] wrote: I think this one must be simple, but i couldn't find it on cisco documentaiton. Is there a special password for telecaster, or should it be cisco? Chad -- Shed those extra pounds with MSN and The Biggest Loser! Learn more.http://biggestloser.msn.com/
[OSL | CCIE_Voice] Clear IP Setting from a 6608
Does anyone know how to do this? Please see the wierdness below. VOICE-6500-1 (enable) sh port 7/4 * = Configured MAC Address # = 802.1X Authenticated Port Name. Port Name Status Vlan Duplex Speed Type - -- -- -- --- 7/4 POD20-PSTN-T1enabled400 full - unknown Port DHCPMAC-Address IP-Address Subnet-Mask --- - --- --- 7/4 enable 00-d0-c0-d3-12-c3 10.20.200.52255.255.255.0 Port Call-Manager(s) DHCP-Server TFTP-Server Gateway - --- --- --- 7/4 - 10.20.200.2110.20.200.2110.20.200.3 Port DNS-Server(s) Domain - - 7/4 - - Port CallManagerState DSP-Type 7/4 notregisteredC549 VOICE-6500-1 (enable) sh port 7/5 * = Configured MAC Address # = 802.1X Authenticated Port Name. Port Name Status Vlan Duplex Speed Type - -- -- -- --- 7/5 POD20-CONFBRDG enabled400 full - unknown Port DHCPMAC-Address IP-Address Subnet-Mask --- - --- --- 7/5 enable 00-d0-c0-d3-12-c4 10.6.200.57 255.255.255.0 Port Call-Manager(s) DHCP-Server TFTP-Server Gateway - --- --- --- 7/5 - 10.6.200.21 10.6.200.21 10.6.200.3 Port DNS-Server(s) Domain - - 7/5 - - Port CallManagerState DSP-Type 7/5 notregisteredC549 Port NoiseRegen NonLinearProcessing - -- --- 7/5 - - Port Trap IfIndex - --- 7/5 disabled 78 Port Status ErrDisable ReasonPort ErrDisableTimeout Action on Timeout -- --- -- - 7/5 enabled - Enable No Change Idle Detection -- -- VOICE-6500-1 (enable) sh port 7/6 * = Configured MAC Address # = 802.1X Authenticated Port Name. Port Name Status Vlan Duplex Speed Type - -- -- -- --- 7/6 POD20-TRNSCDRenabled400 full - unknown Port DHCPMAC-Address IP-Address Subnet-Mask --- - --- --- 7/6 enable 00-d0-c0-d3-12-c5 10.5.200.61 255.255.255.0 Port Call-Manager(s) DHCP-Server TFTP-Server Gateway - --- --- --- 7/6 - 10.5.200.21 10.5.200.21 10.5.200.3 Port DNS-Server(s) Domain - - 7/6 - - Port CallManagerState DSP-Type 7/6 notregisteredC549
Re: [OSL | CCIE_Voice] Clear IP Setting from a 6608
TFTP on call manager is active. VOICE-6500-1 (enable) set vlan 1 7/5 Command authorization failed. VOICE-6500-1 (enable) set vlan 1 7/6 Command authorization failed. VOICE-6500-1 (enable) Thanks, Chad On 1/30/08, Vik Malhi [EMAIL PROTECTED] wrote: A trick with the 6608 is to put the port in the default vlan and then back to the real vlan...You also don't have the TFTP service activated so it will not register. set vlan 1 mod/port set vlan 400 mod/port Vik Malhi CCIE Voice Instructor / Developer - IPexpert, Inc. CCIE Voice #13890 CCSI #31584 URL: http://www.IPexpert.com http://www.ipexpert.com/ Toll Free: +1.866.225.8064 International: +1.810.326.1444 -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Chad Stachowicz *Sent:* Wednesday, January 30, 2008 3:24 PM *To:* CCIE Maillist *Subject:* [OSL | CCIE_Voice] Clear IP Setting from a 6608 Does anyone know how to do this? Please see the wierdness below. VOICE-6500-1 (enable) sh port 7/4 * = Configured MAC Address # = 802.1X Authenticated Port Name. Port Name Status Vlan Duplex Speed Type - -- -- -- --- 7/4 POD20-PSTN-T1enabled400 full - unknown Port DHCPMAC-Address IP-Address Subnet-Mask --- - --- --- 7/4 enable 00-d0-c0-d3-12-c3 10.20.200.52255.255.255.0 Port Call-Manager(s) DHCP-Server TFTP-Server Gateway - --- --- --- 7/4 - 10.20.200.2110.20.200.2110.20.200.3 Port DNS-Server(s) Domain - - 7/4 - - Port CallManagerState DSP-Type 7/4 notregisteredC549 VOICE-6500-1 (enable) sh port 7/5 * = Configured MAC Address # = 802.1X Authenticated Port Name. Port Name Status Vlan Duplex Speed Type - -- -- -- --- 7/5 POD20-CONFBRDG enabled400 full - unknown Port DHCPMAC-Address IP-Address Subnet-Mask --- - --- --- 7/5 enable 00-d0-c0-d3-12-c4 10.6.200.57 255.255.255.0 Port Call-Manager(s) DHCP-Server TFTP-Server Gateway - --- --- --- 7/5 - 10.6.200.21 10.6.200.21 10.6.200.3 Port DNS-Server(s) Domain - - 7/5 - - Port CallManagerState DSP-Type 7/5 notregisteredC549 Port NoiseRegen NonLinearProcessing - -- --- 7/5 - - Port Trap IfIndex - --- 7/5 disabled 78 Port Status ErrDisable ReasonPort ErrDisableTimeout Action on Timeout -- --- -- - 7/5 enabled - Enable No Change Idle Detection -- -- VOICE-6500-1 (enable) sh port 7/6 * = Configured MAC Address # = 802.1X Authenticated Port Name. Port Name Status Vlan Duplex Speed Type - -- -- -- --- 7/6 POD20-TRNSCDRenabled400 full - unknown Port DHCPMAC-Address IP-Address Subnet-Mask --- - --- --- 7/6 enable 00-d0-c0-d3-12-c5 10.5.200.61 255.255.255.0 Port Call-Manager(s) DHCP-Server TFTP-Server Gateway - --- --- --- 7/6 - 10.5.200.21 10.5.200.21 10.5.200.3 Port DNS-Server(s) Domain - - 7/6 - - Port CallManagerState DSP-Type 7/6 notregisteredC549
Re: [OSL | CCIE_Voice] IPCC Agent login on UniverCD
Thank dude, I'm getting down to the nitty gritty here and trying to tie up the silly loose ends before my first attempt! Chad On 1/30/08, boonchin .ng [EMAIL PROTECTED] wrote: Well, why not use the Cisco CAD Installation GuideCAD 6.4 for Unified CMhttp://www.cisco.com/univercd/cc/td/doc/product/voice/sw_ap_to/apps_5_0/english/agents/cad64ig.pdfunder Cisco Customer Response Solutions 5.0(x) in UniverCD? On Jan 31, 2008 1:40 PM, Chad Stachowicz [EMAIL PROTECTED] wrote: it looks like the IPCC Express SRND is one of he links broken on UniverCD... does anyone know a good place to locate the ip phone service URL for CCM, other then in UniversCD? Chad
[OSL | CCIE_Voice] IPCC Question
I think this one must be simple, but i couldn't find it on cisco documentaiton. Is there a special password for telecaster, or should it be cisco? Chad
[OSL | CCIE_Voice] odd CME phone registration
Vik/Mark pod 22 P22-BR2-RTR#sh ephone reg ephone-1 Mac:0017.943A.987E TCP socket:[1] activeLine:0 REGISTERED in SCCP ver 6 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 IP:10.22.202.50 49218 Telecaster 7940 keepalive 79 max_line 2 ephone-2 Mac:0007.EB39.B667 TCP socket:[2] activeLine:0 REGISTERED in SCCP ver 6 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 IP:10.22.202.51 49260 Telecaster 7940 keepalive 79 max_line 2 ephone-3 Mac:001B.D4C6.14F5 TCP socket:[3] activeLine:0 REGISTERED in SCCP ver 6 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 IP:10.100.210.101 50435 Telecaster 7960 keepalive 20 max_line 6 ephone-4 Mac:001B.D4C6.CA2A TCP socket:[4] activeLine:0 REGISTERED in SCCP ver 6 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 IP:10.100.210.102 49695 Telecaster 7960 keepalive 20 max_line 6 P22-BR2-RTR# check ephone-3 and ephone-4 these aren't from my pod ;) Chad
[OSL | CCIE_Voice] Cisco Unity Express Script Editor
All, is there somewhere within the pod to dl this from? I'm wondering if they ask this in the lab how I dl the CUE script editor to my desktop.. Chad
Re: [OSL | CCIE_Voice] IPMA application page
yes that was my rendition of emulating IP. it times out for both the application install and managers page. Pod 6. IPMA service was running On Jan 22, 2008 7:56 AM, Mark Snow [EMAIL PROTECTED] wrote: did you replace the IP address in that URI where it says http://server ? Mark Snow CCIE #14073 (Voice, Security) CCSI #31583 Senior Technical Instructor - IPexpert, Inc. A Cisco Learning Partner - We Accept Learning Credits! Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: [EMAIL PROTECTED] IPexpert - The Global Leader in Self-Study, Classroom-Based, Video On Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. On Jan 22, 2008, at 12:20 AM, Chad Stachowicz wrote: Guys, I enabled the IPMA service and ran the qizard, however I canno seem to get any installation script to run when i got to the http://server/ma/Install/IPMAConsoleInstall.jsp location to download the software. any ideas or something I missed? Thanks, Chad
[OSL | CCIE_Voice] IPIPGW registered to gatekeeper
All, I'm a little confused on when to use what types of trunks from ccm - IPIPGW. If I have an IPIPGW registered to my gatekeeper and my dial peer's use session target ras. Should I be sending calls to the IPIPGW with an ICT(Gatekeeper controlled). Thanks for the calrification, Chad
Re: [OSL | CCIE_Voice] Gatekeeper to CCM Registration
by trunk name do you mean the h225 gatekeeper controlled Device Name? Chad On 1/19/08, senthil natarajan [EMAIL PROTECTED] wrote: yes there is. making your endpoints appear static in the GK. create the trunk in the ccm - gktrunk (for example ) which points to the GK. In the ccm service parameters, search for gk ccm service 1 - set the trunk name there. ccm service 2 - hardcode the RAS port to 1720 (instead of letting the cm pick the RAS port, which is a random number in 5 range). set the above parameters and restart the ccm service. in the GK, set the static alias (for the required e164 numbers registered to the ccm). also make sure when you regsiter the static alias of (pub and sub) define them as the gateway (not the terminal) -Senthil On Jan 19, 2008 4:09 PM, Chad Stachowicz [EMAIL PROTECTED] wrote: In my lab last night I couldn't find anyway to send calls to call manager without using a tech prefix under the trunk and ge-type-prefix 1# default technology on the gatekeeper? Is there ANY other way? Chad
Re: [OSL | CCIE_Voice] BR1 and Cat6K showed registered but CCM is NOT
Is that really a testable topic? What is the best way to manually figure out replication is out of sync? Chad Sent from my iPhone On Dec 23, 2007, at 10:26 PM, Mark Snow [EMAIL PROTECTED] wrote: No DBHelper available in the real lab. Delete if there the Subscription to the CCM0301 DB on the Subscriber server, and manually set it back up again. I will have a video up on our webserver soon that you can view on specifically how to manually setup the Pub/Sub pull subscription. Cheers, Mark Snow CCIE #14073 (Voice, Security) CCSI #31583 Senior Technical Instructor - IPexpert, Inc. A Cisco Learning Partner - We Accept Learning Credits! Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: [EMAIL PROTECTED] IPexpert - The Global Leader in Self-Study, Classroom-Based, Video On Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. On Dec 23, 2007, at 1:57 PM, Chad Stachowicz wrote: Mark how are we expected to fix replication ? Dbhelper? Thanks. Chad Sent from my iPhone On Dec 22, 2007, at 1:00 PM, Mark Snow [EMAIL PROTECTED] wrote: When you do a sh port on the 6608 port, what ccm server does it show reg to? Are you sure that SQL replication is fixed between pub and sub? (by default all of our pods are broken on purpose as a task for the student to fix) Mark Snow Sr Technical Instructor IPexpert, Inc. Sent from my iPhone On Dec 22, 2007, at 11:47 AM, Toyin Bakare [EMAIL PROTECTED] wrote: Hello All, I have registered BR1 and Cat6K in the CCM, when i do sh port on cat6k or sh ccm on BR1, it shows status as registered but CCM shows registration as unknown. I have restarted CCM several times, i deleted and re-add also several time. thnks
Re: [OSL | CCIE_Voice] DSP Resources on CME vs Brance Router
I just tried something else, its only on a dspfarm profile X conference transcode profiles seem to be alright. On Dec 11, 2007 9:57 PM, Chad Stachowicz [EMAIL PROTECTED] wrote: I have my resources configured identically on a CME and branch router ( except of course SCCP local and ip address's) on my branch router when I type Maximum Session 0-0 I dont have any available sessions? I have never seen this before where nothing is available to me... Chad
