[OSL | CCIE_Voice] Vol 2 Lab 1 Gatekeeper

2010-05-10 Thread Mad Kiwi
Hi everyone,

I'm having trouble getting the Gatekeeper lab working, Question 4.2

The config is as per the lab, but it's seems that the transcoder isn't being
invoked.

So from CCM -> G.729 -> HQ-RTR (outvia ipipgw) -> G.711 BR2

When I place a call and issue the "show gatek calls" I can see the G.729
call from the CCM and G.711 to BR2 as per the questions picture.

The call rings and I can answer it but no audio, and then a few seconds
later the call is dropped. It's as if the codec can not be negotiated.

I can see that the transcoder is locally registered to telephony-services
using the command "show sdspfarm unit" and the dial-peers both incoming and
destination-pattern are as per the lab.

Any ideas?

-- 
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Re: [OSL | CCIE_Voice] Switch QoS - Vol.2 Lab 8 Q5.3

2010-05-06 Thread Mad Kiwi
Thanks Sergio,

That's what is was struggling to find last night.

Regards

Kiwi

On Fri, May 7, 2010 at 7:32 AM, Sergio Polizer  wrote:

>  Hi,
>
> This is the expected behavior.
>
> According to Catalyst 3750 Switch Software Configuration Guide, 12.2(44)SE:
>
> "Do not use the* show policy-map* *interface* privileged EXEC command to
> display classification information for incoming traffic. The *
> control-plane* and *interface* keywords are not supported, and the
> statistics shown in the display should be ignored. "
>
> HTH, Sergio.
> --
> Date: Thu, 6 May 2010 21:40:17 +1000
> From: ccievoi...@gmail.com
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] Switch QoS - Vol.2 Lab 8 Q5.3
>
>
> Hi everyone,
>
> I'm working through the HQ (3750) Switch QOS lab, where we use AutoQos,
> should I be able to see packet classification under the "show policy-map
> interface" statement?
>
> As far as I can tell straight out of the box Auto QoS isn't matching the
> packets.
>
> Any thoughts?
>
> --
> Cheers,
>
> kiwi
>
> --
> EM 2009 ACONTECERAM 250.362 FRAUDES NA INTERNET. CLIQUE PARA LER DICAS DE
> SEGURANÇA.
>



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[OSL | CCIE_Voice] Switch QoS - Vol.2 Lab 8 Q5.3

2010-05-06 Thread Mad Kiwi
Hi everyone,

I'm working through the HQ (3750) Switch QOS lab, where we use AutoQos,
should I be able to see packet classification under the "show policy-map
interface" statement?

As far as I can tell straight out of the box Auto QoS isn't matching the
packets.

Any thoughts?

-- 
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kiwi
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Re: [OSL | CCIE_Voice] Proctorlabs seems to have gone down

2010-05-05 Thread Mad Kiwi
Hi guys,

Check that you can resolve the website to it's DNS name, there were DNS
changes worldwide on the 5th of May which maybe impacting your connections.

I'm able to connect from Australia to their site.

Cheers,

On Thu, May 6, 2010 at 1:45 PM, vccie2010  wrote:

> Yeah, same here Roger.
>
> On Wed, May 5, 2010 at 8:40 PM, Roger Henderson wrote:
>
>> Hey All,
>>
>> Is anyone else having trouble with Proctorlabs? My session just dropped
>> out and I can't contact the website to seek assistance or connect via VPN.
>>
>> Roger
>>
>> ___
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>> visit www.ipexpert.com
>>
>>
>
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>
>


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Re: [OSL | CCIE_Voice] Voice Vol II labs 6 & 7 - IPCC Express scripts

2010-05-02 Thread Mad Kiwi
Hi all,

The scripts aren't always on the servers - I had the same problem last week
both booked Proctor lab times, neither had the contact centre scripts local,
I found them under the members area on IPExpert.

Cheers,

On Mon, May 3, 2010 at 1:22 PM, Roger Henderson wrote:

> They're on the desktop of the UCCX server in a folder called 'Student".
>
> On Sun, Feb 21, 2010 at 7:19 PM, Dave Wong  wrote:
>
>> Hi
>> I'm having problems with the UCCX section for labs 6 and 7. I can't find
>> the answer scripts ( .aef files) that are used for these sections of the lab
>> in my member's area. I'm wondering if these are actually available for
>> download, or has anyone successfully completed these sections and are
>> willing to post your scripts?
>>
>>
>>
>> ___
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>> visit www.ipexpert.com
>>
>>
>
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>
>


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Re: [OSL | CCIE_Voice] Lab 9a - IPMA CTI Route Point fails to register

2010-04-22 Thread Mad Kiwi
Hi,

Yes I have added the cti route point in the ipma service parameter, on both
servers.

Regards

kiwi

On Thu, Apr 22, 2010 at 8:22 PM, Angel Perez  wrote:

>  Hi:
>
> Did you add the cti route point at ipma service parameter?
>
> You have to do it at both pub and sub server becouse this parameter is not
> clusterwide and apply only to one server
>
> hth
>
> --
> Date: Thu, 22 Apr 2010 19:31:55 +1000
> From: ccievoi...@gmail.com
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] Lab 9a - IPMA CTI Route Point fails to register
>
>
> Hi everyone,
>
> The IPMA / CTI Route Point seem to be setup as per the lab 9a. However
> although the route point do not register. Restarted all servers and
> CTIManager but nothing changes
>
> I've been looking through the CTI trace logs, but nothing stands out
>
> Any tips on what to try/test please ?
>
> --
> Cheers,
>
> kiwi
>
> --
> Hotmail: Powerful Free email with security by Microsoft. Get it 
> now.
>



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[OSL | CCIE_Voice] Lab 9a - IPMA CTI Route Point fails to register

2010-04-22 Thread Mad Kiwi
Hi everyone,

The IPMA / CTI Route Point seem to be setup as per the lab 9a. However
although the route point do not register. Restarted all servers and
CTIManager but nothing changes

I've been looking through the CTI trace logs, but nothing stands out

Any tips on what to try/test please ?

-- 
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kiwi
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[OSL | CCIE_Voice] IPMA Reading - Videos

2010-04-21 Thread Mad Kiwi
Hi everyone,

Does anyone know of any good IPMA reading or training videos, its a product
i've never used before and I'm trying to get a better understanding of lab
9a and the different modes

thanks
-- 
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Re: [OSL | CCIE_Voice] Is there a way to shutdown make one of VIC2-FXS ports out of service to free DSP resources?

2010-04-20 Thread Mad Kiwi
Hi,

No sorry it's not possible to shut-down 1 FXS port.

Cheers,

kiwi

On Tue, Apr 20, 2010 at 11:51 PM, jeremy co  wrote:

> Hi,
>
>
> my router is short on DSP resources, I want to free DSP resources by
> makeing one of two FXS ports out of service.  However, when I shut down one
> port, it will say that both ports are out of service,
>
>
>
>
> Cheers,
>
>
> Jeremy
>
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Re: [OSL | CCIE_Voice] login into CME via the web

2010-04-20 Thread Mad Kiwi
Hi,

What URL are you using ?

http:///telephony_service.html or something else?

Cheers,

kiwi

On Wed, Apr 21, 2010 at 10:52 AM, Omotayo  wrote:

> Hello,
>
> I usually have issue logging into the CUE via GUI.
>
> When i supply the ip add of the loopback and the username and password
> entered under the telephony-service i get Loin into callmanger as
> administrator failed
>
> When i use vlan 400, i get same result
> below is my config
>
> voice service voip
>  allow-connections h323 to h323
>  allow-connections h323 to sip
>  allow-connections sip to h323
>  allow-connections sip to sip
>  sip
>   bind control source-interface Vlan400
>   bind media source-interface Vlan400
> !
>
> ip http server
> ip http authentication local
> no ip http secure-server
> ip http path FLASH:/GUI
>
>
>
> telephony-service
>  sdspfarm units 2
>  sdspfarm transcode sessions 2
>  sdspfarm tag 1 Sitec-conf
>  no privacy
>  conference hardware
>  no auto-reg-ephone
>  max-ephones 5
>  max-dn 10
>  ip source-address 10.10.202.1 port 2000
>  time-zone 42
>  voicemail 3600
>  max-conferences 8 gain -6
>  call-forward pattern .T
>  web admin system name admin password cisco
>  dn-webedit
>  time-webedit
>  transfer-system full-consult
>  transfer-pattern .T
>  create cnf-files version-stamp 7960 Apr 21 2010 08:41:40
> !
>
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> visit www.ipexpert.com
>
>
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Re: [OSL | CCIE_Voice] MoH not working for PSTN

2010-04-20 Thread Mad Kiwi
Hi,

Did you include in route statement in your your loopback interface?

eg.. multicast 239.1.1.1 port 16384 route  

cheers

kiwi

On Wed, Apr 21, 2010 at 7:42 AM, Sam Wilson  wrote:

> Hi,
>
> We have a working MoH for one site and when I tried to use similar
> configuration for a different site, it seems something went missing.
>
> oprtr02#sh ccm-manager music-on-hold
> Current active multicast sessions : 1
>  Multicast   RTP port   Packets   Call   CodecIncoming
>  Address number in/outid  Interface
> ===
> 239.1.1.1 16384   0/0  562  g711ulaw
>
>
> This is the config when a call is placed on hold, now usually we do see the
> numbers incrementing on the Packets In/Out column however it seems that
> there is no activity as its always 0/0.
>
> Any help will be greatly appreciated.
>
> Regards
> Wilson Samuel
>
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[OSL | CCIE_Voice] Access to BACD text file on CME Flash

2010-04-20 Thread Mad Kiwi
Hi everyone,

In a scenario where we have to configure the BACD TCL script does anyone
know if the ReadMe.txt file is included on the flash or do we need to
memorise the full command syntax ?

Regards,

kiwi.
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Re: [OSL | CCIE_Voice] CCIE - new software release

2010-02-10 Thread Mad Kiwi
Hi all,

Any indication if the gateway IOS change?

Thanks,



On Wed, Feb 10, 2010 at 2:33 PM, Jeff Garvas  wrote:

> Ben answered this in post #48 on February 2nd.  Quick re-cap:
>
> CUCM: 7.0(1)
> 3750 IOS:  12.2(44)SE
> 7965 Phones
> CUPC/CIPC: 7.0(1)
>
>
>
> On Tue, Feb 9, 2010 at 9:24 PM, Amy Ryan  wrote:
>
>>
>> Precedence has shown, they could upgrade a minor release for bug fixes,
>> however, they would not test on any newly supported features.  If they were
>> to upgrade to version 8.X, they would have to announce it 6 months in
>> advance.  This will be unlikely in the foreseeable future because they have
>> not yet exhausted the software in this version.
>>
>> Currently the “Ask the Expert – CCIE VOICE” is running on NetPro until
>> February 12, 2010, and you may get more detailed clarification if you post
>> your question there.  The questions are answered by Cisco expert Ben Ng a
>> program manager with the CCIE VOICE program at Cisco Systems.  The link is
>> below:
>>
>> https://supportforums.cisco.com/message/3011648
>>
>> HTH,
>> Amy
>>
>> ---
>> Amy Ryan – CCIE #24677 (Voice)
>> Technical Instructor - IPexpert, Inc.
>> Mailto: *ar...@ipexpert.com
>> *
>> Telephone: +1.810.326.1444
>> Live Assistance, Please visit: www.ipexpert.com/chat <*
>> http://www.ipexpert.com/chat*>
>> eFax: +1.810.454.0130
>>
>> IPexpert is a premier provider of Classroom and Self-Study Cisco CCNA
>> (R&S, Voice & Security), CCNP, CCVP, CCSP and CCIE (R&S, Voice, Security &
>> Service Provider) Certification Training with locations throughout the
>> United States, Europe and Australia. Be sure to check out our online
>> communities at www.ipexpert.com/communities <*
>> http://www.ipexpert.com/communities*>  and our public website at
>> www.ipexpert.com <*http://www.ipexpert.com*> .
>>
>>
>>
>> --
>> *From: *Michael Thompson 
>> *Date: *Tue, 9 Feb 2010 18:32:34 -0500
>> *To: *Saeed IDris 
>> *Cc: *
>> *Subject: *Re: [OSL | CCIE_Voice] CCIE - new software release
>>
>>
>> I would think that the end result would be anything in the 7.X.X train is
>> testable.  I don't forsee them jumping major revisions like that as the
>> additional content available is to be pretty substantial.
>>
>> On Tue, Feb 9, 2010 at 4:54 PM, Saeed IDris  wrote:
>>
>>
>> Hi :
>> First of all I would like to Thanks community support from IP Expert ,  can
>> someone confirm the below line:
>> -  CCIE Voice - Cisco .com:
>>
>> “ *CCIE* *Software Versions:
>> Any major software release which has been generally
>> available for six months is eligible for testing in the
>> CCIE Voice Lab Exam*.”
>>
>> What I understand from the above lines that Cisco know release CUCM 8.0
>> this month which mean it will be testable topic after 6 month. “JULY 2010”,
>> because im plan my lab on July.
>>
>> Regards,
>> SID
>>
>> ___
>> 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
>>
>> ___
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>> visit www.ipexpert.com
>>
>>
>
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Re: [OSL | CCIE_Voice] SIP Phone Codec ?

2010-02-03 Thread Mad Kiwi
That makes sense, I haven't tested voip dial-peers.

On Wed, Feb 3, 2010 at 9:38 PM, Berry, Matthew J.
wrote:

> I went back and reviewed Vik's VoD walkthrough. SIP voice dial peers do not
> support codec negotiation using voice class codec. Calling within a location
> would not utilize dial peers.
>
>
> - Sent from my Blackberry
>
> ------
>  *From*: Mad Kiwi 
> *To*: Sergio Polizer 
> *Cc*: Berry, Matthew J.; ccie_voice@onlinestudylist.com <
> ccie_voice@onlinestudylist.com>
> *Sent*: Wed Feb 03 03:27:15 2010
>
> *Subject*: Re: [OSL | CCIE_Voice] SIP Phone Codec ?
>
> Hi guys,
>
> Yes, each time I entered the create profile, then reset the phones.
>
> Regards,
>
> Mike.
>
> On Tue, Feb 2, 2010 at 10:42 PM, Sergio Polizer wrote:
>
>>  Mike,
>>
>> I'm wondering if you have entered with "create profile" and proceed with a
>> reset sip phone after each test?
>> Thank You,
>>
>> Sergio.
>> --
>> Date: Tue, 2 Feb 2010 14:00:03 +1100
>> From: ccievoi...@gmail.com
>> To: mjbe...@krollontrack.com
>> CC: ccie_voice@onlinestudylist.com
>>
>> Subject: Re: [OSL | CCIE_Voice] SIP Phone Codec ?
>>
>> Hi guys,
>>
>> Ok, I've tested the configuration by removing the G.711ulaw codec from the
>> list in the voice class.
>>
>> X-lite calls fail because it only supports G.711.
>>
>> SIP phone to SCCP phone within the same CME config - BR2 succeed using
>> G.729
>>
>> SIP phone to SIP phone within the same CME config also succeed using G.729
>>
>> I added the G.711ulaw codec back in as preference 2.
>>
>> And what I found was if the G.711ulaw codec is in the preference list, the
>> phones will use this codec over the lower quality G.729 regardless of the
>> preference order.
>>
>>
>> I hope this helps,
>>
>> Cheers,
>>
>> Mike.
>>
>> On Tue, Feb 2, 2010 at 1:33 PM, Berry, Matthew J. <
>> mjbe...@krollontrack.com> wrote:
>>
>> According to Vik's vol 1 walkthrough, that willk not work. CUCME SIP
>> phones support one codec only.
>> - Sent from my Blackberry
>>
>> --
>>  *From*: ccie_voice-boun...@onlinestudylist.com <
>> ccie_voice-boun...@onlinestudylist.com>
>> *To*: Mike Brooks <2xcci...@gmail.com>
>> *Cc*: OSL Group 
>> *Sent*: Mon Feb 01 20:28:55 2010
>> *Subject*: Re: [OSL | CCIE_Voice] SIP Phone Codec ?
>>
>> Hi Mike,
>>
>> Rather than hardcoding the codec under the voice register pool X
>>
>> use the voice-class codec X command in place of the codec
>>
>> for example;
>>
>> voice class codec 1
>>  codec preference 1 g729r8
>>  codec preference 2 g711ulaw
>>
>>
>> voice register pool 1
>>  id mac ..
>>  type 7961
>>  voice-class codec 1
>>
>> This should work, I haven't tested it yet :-)
>>
>>
>>
>>  On Tue, Feb 2, 2010 at 12:55 PM, Mike Brooks <2xcci...@gmail.com> wrote:
>>
>>  Alright, this is an easy question. I just want to verify what I am
>> thinking is correct.
>>
>> On CME the sip phones have a limitation of having to be hardcoded to a
>> codec, for example G711 OR G729.  I believe the fact that on CME the phones
>> cannot negotiate codec is a limitation with CME and not the sip phones. When
>> phones are registered with callmanager they do not have this problem.
>>
>> Please correct me if I am wrong.
>>
>>
>> Thanks,
>> Mike
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>>
>>
>> --
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>> 8.<http://go.microsoft.com/?linkid=9707132>
>>
>
>
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Re: [OSL | CCIE_Voice] SIP Phone Codec ?

2010-02-03 Thread Mad Kiwi
Hi guys,

Yes, each time I entered the create profile, then reset the phones.

Regards,

Mike.

On Tue, Feb 2, 2010 at 10:42 PM, Sergio Polizer wrote:

>  Mike,
>
> I'm wondering if you have entered with "create profile" and proceed with a
> reset sip phone after each test?
> Thank You,
>
> Sergio.
> --
> Date: Tue, 2 Feb 2010 14:00:03 +1100
> From: ccievoi...@gmail.com
> To: mjbe...@krollontrack.com
> CC: ccie_voice@onlinestudylist.com
>
> Subject: Re: [OSL | CCIE_Voice] SIP Phone Codec ?
>
> Hi guys,
>
> Ok, I've tested the configuration by removing the G.711ulaw codec from the
> list in the voice class.
>
> X-lite calls fail because it only supports G.711.
>
> SIP phone to SCCP phone within the same CME config - BR2 succeed using
> G.729
>
> SIP phone to SIP phone within the same CME config also succeed using G.729
>
> I added the G.711ulaw codec back in as preference 2.
>
> And what I found was if the G.711ulaw codec is in the preference list, the
> phones will use this codec over the lower quality G.729 regardless of the
> preference order.
>
>
> I hope this helps,
>
> Cheers,
>
> Mike.
>
> On Tue, Feb 2, 2010 at 1:33 PM, Berry, Matthew J. <
> mjbe...@krollontrack.com> wrote:
>
> According to Vik's vol 1 walkthrough, that willk not work. CUCME SIP phones
> support one codec only.
> - Sent from my Blackberry
>
> --
>  *From*: ccie_voice-boun...@onlinestudylist.com <
> ccie_voice-boun...@onlinestudylist.com>
> *To*: Mike Brooks <2xcci...@gmail.com>
> *Cc*: OSL Group 
> *Sent*: Mon Feb 01 20:28:55 2010
> *Subject*: Re: [OSL | CCIE_Voice] SIP Phone Codec ?
>
> Hi Mike,
>
> Rather than hardcoding the codec under the voice register pool X
>
> use the voice-class codec X command in place of the codec
>
> for example;
>
> voice class codec 1
>  codec preference 1 g729r8
>  codec preference 2 g711ulaw
>
>
> voice register pool 1
>  id mac ..
>  type 7961
>  voice-class codec 1
>
> This should work, I haven't tested it yet :-)
>
>
>
>  On Tue, Feb 2, 2010 at 12:55 PM, Mike Brooks <2xcci...@gmail.com> wrote:
>
>  Alright, this is an easy question. I just want to verify what I am
> thinking is correct.
>
> On CME the sip phones have a limitation of having to be hardcoded to a
> codec, for example G711 OR G729.  I believe the fact that on CME the phones
> cannot negotiate codec is a limitation with CME and not the sip phones. When
> phones are registered with callmanager they do not have this problem.
>
> Please correct me if I am wrong.
>
>
> Thanks,
> Mike
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
>
>
> --
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> 8.
>
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Re: [OSL | CCIE_Voice] SIP Phone Codec ?

2010-02-01 Thread Mad Kiwi
Hi guys,

Ok, I've tested the configuration by removing the G.711ulaw codec from the
list in the voice class.

X-lite calls fail because it only supports G.711.

SIP phone to SCCP phone within the same CME config - BR2 succeed using G.729

SIP phone to SIP phone within the same CME config also succeed using G.729

I added the G.711ulaw codec back in as preference 2.

And what I found was if the G.711ulaw codec is in the preference list, the
phones will use this codec over the lower quality G.729 regardless of the
preference order.


I hope this helps,

Cheers,

Mike.

On Tue, Feb 2, 2010 at 1:33 PM, Berry, Matthew J.
wrote:

> According to Vik's vol 1 walkthrough, that willk not work. CUCME SIP phones
> support one codec only.
> - Sent from my Blackberry
>
> --
>  *From*: ccie_voice-boun...@onlinestudylist.com <
> ccie_voice-boun...@onlinestudylist.com>
> *To*: Mike Brooks <2xcci...@gmail.com>
> *Cc*: OSL Group 
> *Sent*: Mon Feb 01 20:28:55 2010
> *Subject*: Re: [OSL | CCIE_Voice] SIP Phone Codec ?
>
> Hi Mike,
>
> Rather than hardcoding the codec under the voice register pool X
>
> use the voice-class codec X command in place of the codec
>
> for example;
>
> voice class codec 1
>  codec preference 1 g729r8
>  codec preference 2 g711ulaw
>
>
> voice register pool 1
>  id mac ..
>  type 7961
>  voice-class codec 1
>
> This should work, I haven't tested it yet :-)
>
>
>
> On Tue, Feb 2, 2010 at 12:55 PM, Mike Brooks <2xcci...@gmail.com> wrote:
>
>> Alright, this is an easy question. I just want to verify what I am
>> thinking is correct.
>>
>> On CME the sip phones have a limitation of having to be hardcoded to a
>> codec, for example G711 OR G729.  I believe the fact that on CME the phones
>> cannot negotiate codec is a limitation with CME and not the sip phones. When
>> phones are registered with callmanager they do not have this problem.
>>
>> Please correct me if I am wrong.
>>
>>
>> Thanks,
>> Mike
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>
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Re: [OSL | CCIE_Voice] SIP Phone Codec ?

2010-02-01 Thread Mad Kiwi
Hi Mike,

Rather than hardcoding the codec under the voice register pool X

use the voice-class codec X command in place of the codec

for example;

voice class codec 1
 codec preference 1 g729r8
 codec preference 2 g711ulaw


voice register pool 1
 id mac ..
 type 7961
 voice-class codec 1

This should work, I haven't tested it yet :-)



On Tue, Feb 2, 2010 at 12:55 PM, Mike Brooks <2xcci...@gmail.com> wrote:

> Alright, this is an easy question. I just want to verify what I am thinking
> is correct.
>
> On CME the sip phones have a limitation of having to be hardcoded to a
> codec, for example G711 OR G729.  I believe the fact that on CME the phones
> cannot negotiate codec is a limitation with CME and not the sip phones. When
> phones are registered with callmanager they do not have this problem.
>
> Please correct me if I am wrong.
>
>
> Thanks,
> Mike
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
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Re: [OSL | CCIE_Voice] NM-ESW-16 PoE 7945 phone

2010-01-30 Thread Mad Kiwi
Hi,

There should be no difference between the 7945 and older phones except the
power output requirements, I'm sure that i've powered up 7975's on the
NM-ESW-16 in the past which are class-3 devices.

Are you able to power up an older phone using the same switchport?

If not, check that you have an IPTel power supply installed in your router -
check the telephone symbol next to the power switch.

Cheers,

On Sun, Jan 31, 2010 at 8:38 AM, bontacommunications <
bontacommunicati...@gmail.com> wrote:

> I am running into an issue with bringing up a new 7945 in my lab rack -
> 2811 w/ NM-ESW-16 w/ PoE card
>
> Has anyone successfully powered up a 7945 phone on a NM-ESW-16 w/PoE?
>
> What did you do?
>
>
>
> Thanks in Advance,
> Chris
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
___
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Re: [OSL | CCIE_Voice] Any difference between 7962 and 7965 for lab study?

2010-01-11 Thread Mad Kiwi
That's correct, there is no functional difference between the 2 phones.

In my lab I'm using 7961's, the key thing is the multi-state led for the BLF
status.


On Tue, Jan 12, 2010 at 1:05 PM, Scott ODonnell wrote:

> I have a couple 7962's but no 7965's.
>
> Looking at the technical specs, they look the same other than the color
> display.
>
> Is this correct?
>
>
>
> ___
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> visit www.ipexpert.com
>
>
___
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Re: [OSL | CCIE_Voice] UCM AND CUE INTEGRATION

2010-01-07 Thread Mad Kiwi
Hi,

Not that I've looked at Lab 4 yet (i'm assuming the second workbook) but you
need to use G.711 as CTI Ports do not support G.729.

Regards,

Mike.

On Thu, Jan 7, 2010 at 8:40 PM, Omotayo  wrote:

> Hello All,
>
> Am working on lab 4
> i have integrated the UCM to CUE. Everything shows registered
>
> end
> CUE# show ccn status ccm-manager
> JTAPI Subsystem is currently registered with Call Manager: 10.10.210.10
> JTAPI Version: 7.0(1.1000) -1 Release
>
> when i dial the pilot number 3600, i get a busy tone
>
> when i dial the CTI Port, it gives Not Enough Bandwidth.
>
> I suspect its the location configurtion affecting it . i have 48 configured
> for the branch 2 location has requested by the question to allow just 2 g729
> calls between locations
>
> When i changed the bandwidth to 96. it worked
>
> Anyone with an explanation for this behavouir
> REgards
>
> ___
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> visit www.ipexpert.com
>
>
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Re: [OSL | CCIE_Voice] Codec Negotiation for CTI Ports for IPCC Mobile Agents

2010-01-06 Thread Mad Kiwi
Hi,

You could use Cisco CUBE (IPIPGW) to transcode and route the calls to the
GK.

CCM G.711 -> CUBE with DSP XCODE -> GK G.729

To retain the call supplementary features when integrating CCM to a
gatekeeper you need lot of MTP resources or use G.711, if your using G.711
from CCM to GK and then GK to the remote locations using G.729 then the call
needs to be transcoded using CUBE.

CUBE can be placed on the same hardware as the gatekeeper, dialpeers are
used to control the in and outputs from the CCM trunk and to the GK.

I think one of the IPExpert lab configurations so a sample of a call
transcoded via CUBE to a gatekeeper.

Regards

Mike.

On Thu, Jan 7, 2010 at 4:26 AM, Mann Chaddha  wrote:

> Hi
>
> As per Cisco's Doc, LCP/RCP CTI Ports which are required for Mobile
> Agents CANNOT negotiate Codec and need to have same Codec for Inbound
> & Outbound Calls. I have a scenario where the remote users (Mobile
> Agents) reside on another CUCM Cluster. The calls over the 2 clusters
> shall be G729 as they are over the WAN. The Clusters have H323 Trunks
> with GK for Routing purpose.
>
> Has anyone of you faced a similar limitation & how did you avoid it?
> Is there a way you can invoke Xcoders for CTI Ports ( Jtapi Gateway on
> PG does patch the media between Inbound & Outbound Call Legs)?
>
> I think this is great forum for this question, though IPCCE ain't on
> CCIE blueprint.
>
> Thanks
> Mann
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> visit www.ipexpert.com
>
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