Re: [OSL | CCIE_Voice] workbook 2 lab 8

[Cristobal Priego]

thank you guys
i really appreciate it

2011/4/23 George Goglidze 

> On CUE you most probably have a mismatch in what you've bind sip control ip
> to in voice service VoIP,  and on cue "sip server" by default is the default
> gateway.
>
> Sent from my iPad
>
> On 23 Apr 2011, at 04:56, Cristobal Priego 
> wrote:
>
> Hello all
>
> I just finished my session and i was doing lab 8 from workbooks 2
> something that i couldn't get to work properly was
>
> the unity integration through sip trunk
> the ring no answer and the busy were playing "enter your id followed by
> pound" instead of "sorry extension ... is not available record
> your message at the tone"
> i followed the proctor-guide and still it' didn't work
>
> also i couldn't get the MWI to work on CUE sip phones only
>
> my config looked like this
>
> voice register dn 1
> number 3002
> call-forward b2bua busy 3600
> call-forward b2bua noan 3600 timeout 12
> mwi
> name br2 phone 4
>
>
>
> sip-ua
> mwi-server ipv4:10.10.202.2
>
>
>
> i had the unsolicited notify enabled on the cue gui
>
> when i was doing a refresh of the mwi i saw unity express trying to ring my
> extensions on the default mwi extensions
> so i went ahead and configured the ephone dn's for mwi
>
> still didn't work
>
>
> also my sip srst didn't work
>
> i kept getting this error
>
>
>
> Apr 23 07:46:38.660: //-1//SIP/Msg/ccsipDisplayMsg:
> Sent:
> SIP/2.0 404 Not Found
> Via: SIP/2.0/UDP 10.10.202.1:5060;branch=z9hG4bKe43aa9be
> From: 1002@10.10.201.1
> >;tag=001ae22b12c500077cb0194d-395f52ee
> To: 1002@10.10.201.1>;tag=17DB834-A33
> Date: Sat, 23 Apr 2011 07:46:38 GMT
> Call-ID: <001ae22b-12c50006-85a48968-9fe91895@192.168.11.12>
> 001ae22b-12c50006-85a48968-9fe91895@192.168.11.12
> Server: Cisco-SIPGateway/IOS-12.x
> CSeq: 105 REGISTER
> Content-Length: 0
>
>
> my sip srst looked like this
>
> voice register pool 1
> id network 10.10.201.0 mask 255.255.255.0
> cor incoming ld-css default
> call-forward b2bua busy 5600
> call-foward b2bua noan 5600 timeout 13
> codec g711u
>
>
> voice register global
> max-pool 2
> max-dn 2
>
>
>
> please help me out,  thank you
>
>
> ___
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> visit www.ipexpert.com
>
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> www.PlatinumPlacement.com
>
>
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Re: [OSL | CCIE_Voice] MVA using mgcp gateway

[George Goglidze]

 
What do you mean by you call 5999 and CUCM will automatically hairpin the call 
to 12345?
CUCM will not automatically know what you want to do, otherwise ccie's would 
not be needed :)

You'll have to put either translation patter from 5999 to 12345, or a route 
pattern of 5999. 

Basically when the call comes in to 5999 on MGCP gateway, this gateway should 
have a CSS that points to a partition that contains 5999 route point that goes 
to the h323 gateway and which invokes MVA.

Sent from my iPad

On 23 Apr 2011, at 17:23, donny f  wrote:

> i did, if i call 12345678 to vxml, it call the cisco lady.  so go to h323 ok
>  
> the thing is , when i call 5999, it did do hairpin at all  to h323  12345678
>  
> MVA (media grup) : 5999
> MVA (serv parameter) : 1234567
>  
>  
> I assume flow is :   call from PSTN to 5999, and then it will hairpin to H323 
>  1234567 (thru RP in UCM)
>CSS and partition, i have checked
>  
> This is what i did:
> -
> 1. application service CCM
> •http://:8080/ccmivr/pages/IVRMainpage.vxml
> 
> 2. dial-peer voice 1234567 voip
> •service CCM
> 
> •incoming called-number 1234567
> 
> •codec g711u
> 
> •session target ipv4:
> 
>  
> 
> 3.
> 
> •dial-peer voice 101 voip
> 
> •preference 1
> 
> •destination-pattern 
> 
> session target ipv4:10.1.30.3
> 
> •voice-class h323 1
> 
> •codec g711ulaw
> 
> •dtmf-relay h245-alphanumeric
> 
> •no vad
> 
> 4. voice service voip
> 
> •allow-connections h323 to h323
> 
> 5. RP in UCM point to h323 GW to get the  VXML  --- and this work
> 
>  
> 
> Just it do not hairpin to  RP-- VXML  from   5999
> 
> On Sat, Apr 23, 2011 at 3:04 AM, George Goglidze  wrote:
> What troubleshooting have you done? Does the call get to h323 gateway after 
> coming in from MGCP? Does it match correct dial-peer? Does it invoke 
> application? Is application correctly defined? 
> 
> Any debugs done?
> 
> Debug voice dialpeer
> This is to start with... If nothing shows up it means you are not sending a 
> call to h323 gw at all, so check you MGCP gw CSS and route pattern.
> 
> Sent from my iPad
> 
> On 23 Apr 2011, at 05:57, donny f  wrote:
> 
>> hi experts,
>>  
>> I am configureing MVA  MGCP using haipin and following this document.  (Saw 
>> from thread, Roger and Shingei ever make it worked)
>> https://supportforums.cisco.com/thread/2005673
>>  
>> The call to MVA (3456) never hit the MVA  , did i miss anything in this 
>> steps here?
>>  
>> mind advice
>>  
>> tks in adv
> 
>> ___
>> For more information regarding industry leading CCIE Lab training, please 
>> visit www.ipexpert.com
>> 
>> Are you a CCNP or CCIE and looking for a job? Check out 
>> www.PlatinumPlacement.com
> 
___
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Re: [OSL | CCIE_Voice] Marking Traffic on the HQ-Switch

[George Goglidze]

Hi Sam,

Can you check for me if "mls qos" is globally enabled?
You need this on a switch to do any QoS operation... Otherwise your 
configuration looks fine, as long as it's applied to all corresponding 
CUCM,router and phone ports.

Sent from my iPad

On 23 Apr 2011, at 16:40, Sam Park  wrote:

> All;
> 
> I'm looking at the QoS section 5.2 of Lab3 from the OWLE class (for those of 
> you who took it and have the workbook)
> And Vik if you read this, the IPExpert Solution guide you gave us does NOT 
> have the answer to this question.
> 
> Here is the scenario:
> 
> Ensure that all SCCP traffic passing through the CAT 3750 is marked to DSCP 
> AF31.  You can't trust the devices attached to the switchports to do this.
> 
> Now, I went through CUCM and CUC and change SCCP related stuff to AF31.  
> However, when I looked at the traffic in and out of the ports, the phone 
> shows it sends AF31 packets, but CUCM is still seemed to be sending "all" 
> control messages on CS3; meaning I don't see a hint of AF31 traffic.   BTW 
> you can use the command, "show mls qos interface fa0/5 statistics" to see 
> what dscp and cos packets are coming in and out of that particular interface.
> 
> So I tried to use an ip access-list, class-map, and policy-map to classify 
> the traffic and remark it to AF31, and applied the service policy to incoming 
> traffic on the phone ports and Server ports.  But no go.
> Config on the switch as follows
> * Begin
> ip access-list extended SCCP-CONTROL
>  permit tcp any any range 2000 2002
>  permit tcp any eq 2000 any
> 
> class-map match-any SCCP-SIG
>  match access-group name SCCP-CONTROL
> 
> policy-map REMARK-SCCP
>  class SCCP-SIG
>   set dscp af31
> 
> interface FastEthernet0/6
>  description *** HQ Phone 7965 ***
>  switchport access vlan 10
>  switchport mode access
>  switchport voice vlan 20
>  spanning-tree portfast
>  service-policy input REMARK-SCCP
> 
> * End
> 
> Then I used Wireshark on back of the phone, I found that messages from CUCM 
> to the phone actually has DSCP set to 0.  While the phone has the AF31 config 
> from the Enterprise Parameters.  It is using the right port 2000 (src or dest 
> I got that covered in ip access list).  I've restarted the CallManager 
> Services, rebooted the switch, disabled / reenabled mls qos; and I can't get 
> it work.
> I don't see any traffic being classified or remarked as I have configured the 
> policy - a "show policy-map interface" shows no matches on the traffic.  What 
> gives?
> 
> Scratching my head.
> 
> Maybe its my gear.
> I'm using WS-C3560-24PS-E on 12.2(50)SE1
> 
> Wondering if anyone can answer or comment on this? 
> Maybe I should just abandone the 3 points.
> 
> Thanks
> 
> ___
> For more information regarding industry leading CCIE Lab training, please 
> visit www.ipexpert.com
> 
> Are you a CCNP or CCIE and looking for a job? Check out 
> www.PlatinumPlacement.com
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Re: [OSL | CCIE_Voice] MVA using mgcp gateway

[donny f]

i did, if i call 12345678 to vxml, it call the cisco lady.  so go to h323 ok

the thing is , when i call 5999, it did do hairpin at all  to h323  12345678

MVA (media grup) : 5999
MVA (serv parameter) : 1234567


I assume flow is :   call from PSTN to 5999, and then it will hairpin to
H323  1234567 (thru RP in UCM)
   CSS and partition, i have checked

This is what i did:
-
1. application service CCM

•http://:8080/ccmivr/pages/IVRMainpage.vxml
2. dial-peer voice 1234567 voip

•service CCM

•incoming called-number 1234567

•codec g711u

•session target ipv4:



3.

•dial-peer voice 101 voip

•preference 1

•destination-pattern 

session target ipv4:10.1.30.3

•voice-class h323 1

•codec g711ulaw

•dtmf-relay h245-alphanumeric

•no vad

4. voice service voip

•allow-connections h323 to h323

5. RP in UCM point to h323 GW to get the  VXML  --- and this work



Just it do not hairpin to  RP-- VXML  from   5999
On Sat, Apr 23, 2011 at 3:04 AM, George Goglidze  wrote:

>  What troubleshooting have you done? Does the call get to h323 gateway
> after coming in from MGCP? Does it match correct dial-peer? Does it invoke
> application? Is application correctly defined?
>
> Any debugs done?
>
> Debug voice dialpeer
> This is to start with... If nothing shows up it means you are not sending a
> call to h323 gw at all, so check you MGCP gw CSS and route pattern.
>
> Sent from my iPad
>
> On 23 Apr 2011, at 05:57, donny f  wrote:
>
>   hi experts,
>
> I am configureing MVA  MGCP using haipin and following this document.  (Saw
> from thread, Roger and Shingei ever make it worked)
> https://supportforums.cisco.com/thread/2005673
>
> The call to MVA (3456) never hit the MVA  , did i miss anything in this
> steps here?
>
> mind advice
>
> tks in adv
>
>  ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
> 
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

[OSL | CCIE_Voice] Marking Traffic on the HQ-Switch

[Sam Park]

All;

I'm looking at the QoS section 5.2 of Lab3 from the OWLE class (for those of
you who took it and have the workbook)
And Vik if you read this, the IPExpert Solution guide you gave us does NOT
have the answer to this question.

Here is the scenario:

Ensure that all SCCP traffic passing through the CAT 3750 is marked to DSCP
AF31.  You can't trust the devices attached to the switchports to do this.

Now, I went through CUCM and CUC and change SCCP related stuff to AF31.
However, when I looked at the traffic in and out of the ports, the phone
shows it sends AF31 packets, but CUCM is still seemed to be sending "all"
control messages on CS3; meaning I don't see a hint of AF31 traffic.   BTW
you can use the command, "show mls qos interface fa0/5 statistics" to see
what dscp and cos packets are coming in and out of that particular
interface.

So I tried to use an ip access-list, class-map, and policy-map to classify
the traffic and remark it to AF31, and applied the service policy to
incoming traffic on the phone ports and Server ports.  But no go.
Config on the switch as follows
* Begin
ip access-list extended SCCP-CONTROL
 permit tcp any any range 2000 2002
 permit tcp any eq 2000 any

class-map match-any SCCP-SIG
 match access-group name SCCP-CONTROL

policy-map REMARK-SCCP
 class SCCP-SIG
  set dscp af31

interface FastEthernet0/6
 description *** HQ Phone 7965 ***
 switchport access vlan 10
 switchport mode access
 switchport voice vlan 20
 spanning-tree portfast
 service-policy input REMARK-SCCP

* End

Then I used Wireshark on back of the phone, I found that messages from CUCM
to the phone actually has DSCP set to 0.  While the phone has the AF31
config from the Enterprise Parameters.  It is using the right port 2000 (src
or dest I got that covered in ip access list).  I've restarted the
CallManager Services, rebooted the switch, disabled / reenabled mls qos; and
I can't get it work.
I don't see any traffic being classified or remarked as I have configured
the policy - a "show policy-map interface" shows no matches on the traffic.
What gives?

Scratching my head.

Maybe its my gear.
I'm using WS-C3560-24PS-E on 12.2(50)SE1

Wondering if anyone can answer or comment on this?
Maybe I should just abandone the 3 points.

Thanks
___
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www.ipexpert.com

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www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] pstn call error

[donny f]

hmm i will try again in today sessionm tks G

On Sat, Apr 23, 2011 at 2:51 AM, George Goglidze  wrote:

>  Looks like you've configured more channels than you're allowed to use.
>
> Sent from my iPad
>
> On 23 Apr 2011, at 05:30, donny f  wrote:
>
>  i config to mgcp and work fine. what i might miss here?
>
> On Fri, Apr 22, 2011 at 7:41 PM, Hough, Earl <
> earl.ho...@pcmallservices.com> wrote:
>
>>  The reason for this error message is not due to H323.  Look at the
>> source of the messages.  They are Q931 messages.
>>
>>
>>
>> What might cause the required circuit/channel not to be available on a
>> PRI?  Hint: look at your base ISDN configuration and compare that to the
>> PSTN emulator for the same circuit.
>>
>>
>>
>>
>>
>> Earl Hough
>>
>> CCIE #16508 (R&S/Security/Voice)
>>
>>
>>
>> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
>> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *donny f
>> *Sent:* Friday, April 22, 2011 8:48 PM
>> *To:* ccie_voice@onlinestudylist.com
>> *Subject:* [OSL | CCIE_Voice] pstn call error
>>
>>
>>
>> hi all,
>>
>>
>>
>> what i miss in h323 config, that cause this ?
>>
>>
>>
>> debug from HQ (h 323)
>>
>>
>>
>> --
>>
>>  Cause i = 0x80AC - Requested circuit/channel not available
>> *Apr 23 05:41:45.559: ISDN Se0/0/0:23 Q931: RX <- RELEASE pd = 8  callref
>> = 0x8087
>>
>> ---
>>
>>
>>
>>
>>
>> deug in PSTN
>>
>> ---
>>
>>   Cause i = 0x80AC - Requested circuit/channel not available
>> Apr 23 04:42:28.023: ISDN Se0/3/0:23 Q931: TX -> RELEASE pd = 8  callref =
>> 0x8087
>> Apr 23 04:42:28.031: ISDN Se0/3/0:23 Q931: RX <- RELEASE_COMP pd = 8
>> callref = 0x0087
>> PSTN-WAN(config-controller)#
>> Apr 23 04:42:56.107: %ENVMON-3-FAN_FAILED: Fan 1 not rotating
>>
>> _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _
>>
>> The information contained in this transmission is confidential. It is
>> intended solely for the use of the individual(s) or organization(s) to
>> whom it is addressed. Any disclosure, copying or further distribution is
>> not permitted unless such privilege is explicitly granted in writing by
>> PC Mall, Inc. Furthermore, PC Mall, Inc. is not responsible for
>> the proper and complete transmission of the substance of this
>> communication, nor for any delay in its receipt.
>>
>>
>>
>  ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
> 
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] Meet Me + Ad-hoc Participant (WB2 Lab1 6.2)

[Alex Goh]

Hi Claude,

Thanks for the reply, I do have the hardware conference bridge configured on
the router, with support of
G729 codec of course. That verify by BR1 Phones can join into meet me
conference by dialing the number.

Issue come in when one of the meetme participant trying bring in another
participant on different codec
region into the bridge.

Thanks

Regards,
Alex

On Sat, Apr 23, 2011 at 6:59 PM, Friderich Claude wrote:

>  Hello,
>
>
>
> I think you are wrong for this question.
>
> You must invoke a conference bridge on the router and thanks to the
> hardware conference bridge it will support g729
>
>
>
> Just add the codec g729r8 in the dspfarm profile conference ….
>
>
>
> Should work.
>
>
>
> Of course do not forget to put your conference bridge in the MRG and MRGL
> of your CCM
>
>
>
> Regards
>
>
>
> Claude.
>
>
>
> *Claude Friderich*
>
> *PreSales Support*
>
> *[image: ccvp_voice_sm]***
>
> *NETCORE PSF S.A.***
>
> 49 rue du Baerendall
>
> B.P.65 L-8201 Mamer
>
> Téléphone: 31 33 80-407
>
> Fax: 31 33 80 8-407
>
> GSM: 621 303 616
>
> E-mail: cfrider...@netcore.lu
>
>
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Alex Goh
> *Sent:* samedi 23 avril 2011 12:10
> *To:* OSL
> *Subject:* [OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 6.2)
>
>
>
> Hi All,
>
> I'm practicing for the question 6.2 in Workbook 2, Lab 1. I manage to have
> the MeetMe conference setup successfuly,
> HQ Phn, BR1 & PSTN all can dial into MeetMe number too. The H/W Conference
> Bridge was configured in HQ-RTR,
> and I can verified it was utilized.
>
> The problem comes in when I try to add an ad-hoc participant into the
> MeetMe Conference Bridge on different region.
> e.g when MeetMe conference Initiator is HQ PH1, and HQ PH2 joined ad-hoc
> participant BR1 PH1 which is on G729,
> the BR1 PH1 will get dropped.
>
> I know this is due to codec mismatch issue (verified by changing region
> from HQ to BR1 as G711, it works fine), but I've
> transcoder added in both HQ & BR1 DP MRGL. It looks like the transcoder
> doesn't get invoked in this case or do transcoder
> needed to get this working? since I already have H/W Conference configured.
>
> Appreciate if anyone can shed more light on this.
>
> Thanks
>
> Regards,
> Alex
>
>
>
> --
>
> This email was Anti Virus checked.
>
>
___
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Re: [OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 6.2)

[Friderich Claude]

Hello,
 
I think you are wrong for this question.
You must invoke a conference bridge on the router and thanks to the hardware 
conference bridge it will support g729
 
Just add the codec g729r8 in the dspfarm profile conference ….
 
Should work.
 
Of course do not forget to put your conference bridge in the MRG and MRGL of 
your CCM
 
Regards
 
Claude.
 
Claude Friderich
PreSales Support
 
NETCORE PSF S.A.
49 rue du Baerendall
B.P.65 L-8201 Mamer
Téléphone: 31 33 80-407
Fax: 31 33 80 8-407
GSM: 621 303 616
E-mail: cfrider...@netcore.lu  
 
From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Alex Goh
Sent: samedi 23 avril 2011 12:10
To: OSL
Subject: [OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 6.2)
 
Hi All,

I'm practicing for the question 6.2 in Workbook 2, Lab 1. I manage to have the 
MeetMe conference setup successfuly,
HQ Phn, BR1 & PSTN all can dial into MeetMe number too. The H/W Conference 
Bridge was configured in HQ-RTR, 
and I can verified it was utilized.

The problem comes in when I try to add an ad-hoc participant into the MeetMe 
Conference Bridge on different region.
e.g when MeetMe conference Initiator is HQ PH1, and HQ PH2 joined ad-hoc 
participant BR1 PH1 which is on G729,
the BR1 PH1 will get dropped.

I know this is due to codec mismatch issue (verified by changing region from HQ 
to BR1 as G711, it works fine), but I've
transcoder added in both HQ & BR1 DP MRGL. It looks like the transcoder doesn't 
get invoked in this case or do transcoder
needed to get this working? since I already have H/W Conference configured.

Appreciate if anyone can shed more light on this.

Thanks

Regards,
Alex
 
-- 
This email was Anti Virus checked. 
<>___
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[OSL | CCIE_Voice] Gatekeeper + CUBE (WB2 Lab1 6.2)

[Alex Goh]

Hi All,

I'm practicing for the question 6.2 in Workbook 2, Lab 1. I manage to have
the MeetMe conference setup successfuly,
HQ Phn, BR1 & PSTN all can dial into MeetMe number too. The H/W Conference
Bridge was configured in HQ-RTR,
and I can verified it was utilized.

The problem comes in when I try to add an ad-hoc participant into the MeetMe
Conference Bridge on different region.
e.g when MeetMe conference Initiator is HQ PH1, and HQ PH2 joined ad-hoc
participant BR1 PH1 which is on G729,
the BR1 PH1 will get dropped.

I know this is due to codec mismatch issue (verified by changing region from
HQ to BR1 as G711, it works fine), but I've
transcoder added in both HQ & BR1 DP MRGL. It looks like the transcoder
doesn't get invoked in this case or do transcoder
needed to get this working? since I already have H/W Conference configured.

Appreciate if anyone can shed more light on this.

Thanks

Regards,
Alex
___
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[OSL | CCIE_Voice] graded lab voice

[Royal Ccie]

Hi,

I have CBootCamp's CCIE Voice Advance Lab workbook ver3.1-lab-1-5 and need 6-10 
labs.. cisco 360 labs...
Yes, anything provided by cisco is good.
I'll share please inform me.

thanks..





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Sent: Sat, April 23, 2011 3:08:32 AM
Subject: CCIE_Voice Digest, Vol 62, Issue 130

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Today's Topics:

  1. Re: graded lab voice (adam compton)
  2. Re: graded lab voice (Joshua Reola)
  3. Re: cue jtapi connection (George Goglidze)
  4. Re: CME busy-trigger-button Problems (Shrini)


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Message: 1
Date: Fri, 22 Apr 2011 14:06:48 -0400
From: adam compton 
To: ee 55 
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] graded lab voice
Message-ID: 
Content-Type: text/plain; charset="iso-8859-1"

How do you purchase that?  Do you know the price?

Adam Compton

On Fri, Apr 22, 2011 at 11:28 AM, ee 55  wrote:

> hi ,
>
> is anybody used cisco voice 360 graded lab before ?  just wondering ,worth
> to spend money on ?
>
> tks
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com 
>
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Message: 2
Date: Fri, 22 Apr 2011 14:25:41 -0500
From: Joshua Reola 
To: adam compton 
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] graded lab voice
Message-ID: 
Content-Type: text/plain; charset="iso-8859-1"

It depends what partner and packages you get.
https://learningnetwork.cisco.com/docs/DOC-7998

If anyone is interested in purchasing as a group let me know. We can get a
discount the larger the group, the better. You can find more info on my
blog.

JR
www.joshreola.com

On Fri, Apr 22, 2011 at 1:06 PM, adam compton  wrote:

> How do you purchase that?  Do you know the price?
>
> Adam Compton
>
> On Fri, Apr 22, 2011 at 11:28 AM, ee 55  wrote:
>
>> hi ,
>>
>> is anybody used cisco voice 360 graded lab before ?  just wondering ,worth
>> to spend money on ?
>>
>> tks
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com 
>>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
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Message: 3
Date: Sat, 23 Apr 2011 00:15:05 +0100
From: George Goglidze 
To: ee 55 
Cc: "ccie_voice@onlinestudylist.com" 
Subject: Re: [OSL | CCIE_Voice] cue jtapi connection
Message-ID: 
Content-Type: text/plain; charset="us-ascii"

Probably outcall is the easiest, although it has some limitations as you can't 
use it with CUCM, and neither for SIP Phones on CUCME.

If you have only SCCP CME, then feel free to use outcall... As for CUCM you 
must 
use unsolicited notify, as CUCM has no way of subscribing for MWI notifications 
to CUE... Or any other voicemail system as a matter of fact.

But for CUCME, for sip or SCCP phones, you can use both, sip notify or 
unsolicited. Does not really matter. You could even have unsolicited for CUCM, 
and on fallback sip notify. Will work just fine as long as it's correctly 
configured.

It's not really about which one is better, it's about the requirements, you 
have 
to know how to identify which one you need depending on the requirements. For 
example if they say, make sure no "invite" is sent on MWI, you should know that 
you can't use outcall...

Regards,

Sent from my iPad

On 22 Apr 2011, at 16:18, ee 55  wrote:

> hi expert,
>  
> what is different in all 3 mwi method : unsolicited, notify, and outcall, 
> from 
>the usage perspective in exam?    which is the safest way
>  
> are the sub-notify and outcall will not work in SRST mode ? mean red light is 
>not lit.
>  
> tks
> 
> On Fri, Apr 22, 2011 at 12:11 AM, George Goglidze  wrote:
> Hi Erwan,
> 
> W

Re: [OSL | CCIE_Voice] cue jtapi connection

[George Goglidze]

I think it should, but I'd have to test. It does have mwi sip-server  
configuration under call-manager-fallback, and  it then tells you it will be 
automatically converted to 
sip-ua
 mwi-server x.x.x.x

Regards,

Sent from my iPad

On 23 Apr 2011, at 05:49, donny f  wrote:

>  
>  
> just to clarify, so   sub-notify and  unsolicited  are work in all scenario, 
> such as : Jtapi CUE  in srst mode , SRST  (call-manager fallback , telephony 
> service)
>  
> meaning they will lit the red light , if we leave message in SRST mode above ?
>  
> 
> 
>  
> On Fri, Apr 22, 2011 at 5:15 PM, George Goglidze  wrote:
> Probably outcall is the easiest, although it has some limitations as you 
> can't use it with CUCM, and neither for SIP Phones on CUCME.
>  
> If you have only SCCP CME, then feel free to use outcall... As for CUCM you 
> must use unsolicited notify, as CUCM has no way of subscribing for MWI 
> notifications to CUE... Or any other voicemail system as a matter of fact.
> 
> But for CUCME, for sip or SCCP phones, you can use both, sip notify or 
> unsolicited. Does not really matter. You could even have unsolicited for 
> CUCM, and on fallback sip notify. Will work just fine as long as it's 
> correctly configured.
> 
> It's not really about which one is better, it's about the requirements, you 
> have to know how to identify which one you need depending on the 
> requirements. For example if they say, make sure no "invite" is sent on MWI, 
> you should know that you can't use outcall...
> 
> Regards,
> 
> Sent from my iPad
> 
> On 22 Apr 2011, at 16:18, ee 55  wrote:
> 
>> hi expert,
>>  
>> what is different in all 3 mwi method : unsolicited, notify, and outcall, 
>> from the usage perspective in exam?which is the safest way
>>  
>> are the sub-notify and outcall will not work in SRST mode ? mean red light 
>> is not lit.
>>  
>> tks
>> 
>> On Fri, Apr 22, 2011 at 12:11 AM, George Goglidze  wrote:
>> Hi Erwan,
>> 
>> When you integrate CUE with CUCM, you will have unsolicited notify sip MWI 
>> enabled. 
>> And by default if in SRST mode, CUE will send same unsolicited notify 
>> message to it's default gateway too...
>> 
>> So as long as you prepare your router to listen to this SIP messages it'll 
>> be fine:
>> 
>> voice service voip
>> sip
>>  bind control 
>> 
>> sip-ua
>>   mwi-server  unsolicited
>> 
>> This should be enough,
>> 
>> Sent from my iPad
>> 
>> On 22 Apr 2011, at 05:53, Erwan Erwan  wrote:
>> 
>>> hi all,
>>>  
>>> when i use JTAPI  connection for cue  to UCM.
>>> In SRST mode, should I get the  MWI in SRST mode ?
>>>  
>>> If yes, what command I need to get the MWI in SRST ?
>>>  
>>> tks in adv
>> 
>>> ___
>>> For more information regarding industry leading CCIE Lab training, please 
>>> visit www.ipexpert.com
>>> 
>>> Are you a CCNP or CCIE and looking for a job? Check out 
>>> www.PlatinumPlacement.com
>> 
>> ___
>> For more information regarding industry leading CCIE Lab training, please 
>> visit www.ipexpert.com
>> 
>> Are you a CCNP or CCIE and looking for a job? Check out 
>> www.PlatinumPlacement.com
>> 
> 
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] MVA using mgcp gateway

[George Goglidze]

What troubleshooting have you done? Does the call get to h323 gateway after 
coming in from MGCP? Does it match correct dial-peer? Does it invoke 
application? Is application correctly defined? 

Any debugs done?

Debug voice dialpeer
This is to start with... If nothing shows up it means you are not sending a 
call to h323 gw at all, so check you MGCP gw CSS and route pattern.

Sent from my iPad

On 23 Apr 2011, at 05:57, donny f  wrote:

> hi experts,
>  
> I am configureing MVA  MGCP using haipin and following this document.  (Saw 
> from thread, Roger and Shingei ever make it worked)
> https://supportforums.cisco.com/thread/2005673
>  
> The call to MVA (3456) never hit the MVA  , did i miss anything in this steps 
> here?
>  
> mind advice
>  
> tks in adv
> ___
> For more information regarding industry leading CCIE Lab training, please 
> visit www.ipexpert.com
> 
> Are you a CCNP or CCIE and looking for a job? Check out 
> www.PlatinumPlacement.com
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] workbook 2 lab 8

[George Goglidze]

On CUE you most probably have a mismatch in what you've bind sip control ip to 
in voice service VoIP,  and on cue "sip server" by default is the default 
gateway. 

Sent from my iPad

On 23 Apr 2011, at 04:56, Cristobal Priego  wrote:

> Hello all
> 
> I just finished my session and i was doing lab 8 from workbooks 2
> something that i couldn't get to work properly was
> 
> the unity integration through sip trunk
> the ring no answer and the busy were playing "enter your id followed by 
> pound" instead of "sorry extension ... is not available record your message 
> at the tone"
> i followed the proctor-guide and still it' didn't work
> 
> also i couldn't get the MWI to work on CUE sip phones only
> 
> my config looked like this
> 
> voice register dn 1
> number 3002
> call-forward b2bua busy 3600
> call-forward b2bua noan 3600 timeout 12
> mwi
> name br2 phone 4
> 
> 
> 
> sip-ua
> mwi-server ipv4:10.10.202.2
> 
> 
> 
> i had the unsolicited notify enabled on the cue gui
> 
> when i was doing a refresh of the mwi i saw unity express trying to ring my 
> extensions on the default mwi extensions
> so i went ahead and configured the ephone dn's for mwi
> 
> still didn't work
> 
> 
> also my sip srst didn't work
> 
> i kept getting this error
> 
> 
> 
> Apr 23 07:46:38.660: //-1//SIP/Msg/ccsipDisplayMsg:
> Sent: 
> SIP/2.0 404 Not Found
> Via: SIP/2.0/UDP 10.10.202.1:5060;branch=z9hG4bKe43aa9be
> From: ;tag=001ae22b12c500077cb0194d-395f52ee
> To: ;tag=17DB834-A33
> Date: Sat, 23 Apr 2011 07:46:38 GMT
> Call-ID: 001ae22b-12c50006-85a48968-9fe91895@192.168.11.12
> Server: Cisco-SIPGateway/IOS-12.x
> CSeq: 105 REGISTER
> Content-Length: 0
> 
> 
> my sip srst looked like this
> 
> voice register pool 1
> id network 10.10.201.0 mask 255.255.255.0
> cor incoming ld-css default
> call-forward b2bua busy 5600
> call-foward b2bua noan 5600 timeout 13
> codec g711u
> 
> 
> voice register global
> max-pool 2
> max-dn 2
> 
> 
> 
> please help me out,  thank you
> 
> 
> ___
> For more information regarding industry leading CCIE Lab training, please 
> visit www.ipexpert.com
> 
> Are you a CCNP or CCIE and looking for a job? Check out 
> www.PlatinumPlacement.com
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] pstn call error

[George Goglidze]

Looks like you've configured more channels than you're allowed to use.

Sent from my iPad

On 23 Apr 2011, at 05:30, donny f  wrote:

> i config to mgcp and work fine. what i might miss here?
> 
> On Fri, Apr 22, 2011 at 7:41 PM, Hough, Earl  
> wrote:
> The reason for this error message is not due to H323.  Look at the source of 
> the messages.  They are Q931 messages. 
> 
>  
> 
> What might cause the required circuit/channel not to be available on a PRI?  
> Hint: look at your base ISDN configuration and compare that to the PSTN 
> emulator for the same circuit.
> 
>  
> 
>  
> 
> Earl Hough
> 
> CCIE #16508 (R&S/Security/Voice)
> 
>  
> 
> From: ccie_voice-boun...@onlinestudylist.com 
> [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of donny f
> Sent: Friday, April 22, 2011 8:48 PM
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] pstn call error
> 
>  
> 
> hi all,
> 
>  
> 
> what i miss in h323 config, that cause this ?
> 
>  
> 
> debug from HQ (h 323)
> 
>  
> 
> --
> 
>  Cause i = 0x80AC - Requested circuit/channel not available
> *Apr 23 05:41:45.559: ISDN Se0/0/0:23 Q931: RX <- RELEASE pd = 8  callref = 
> 0x8087
> 
> ---
> 
>  
> 
>  
> 
> deug in PSTN
> 
> ---
> 
>   Cause i = 0x80AC - Requested circuit/channel not available
> Apr 23 04:42:28.023: ISDN Se0/3/0:23 Q931: TX -> RELEASE pd = 8  callref = 
> 0x8087
> Apr 23 04:42:28.031: ISDN Se0/3/0:23 Q931: RX <- RELEASE_COMP pd = 8  callref 
> = 0x0087
> PSTN-WAN(config-controller)#
> Apr 23 04:42:56.107: %ENVMON-3-FAN_FAILED: Fan 1 not rotating
> 
> _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _
> 
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> 
> 
> ___
> For more information regarding industry leading CCIE Lab training, please 
> visit www.ipexpert.com
> 
> Are you a CCNP or CCIE and looking for a job? Check out 
> www.PlatinumPlacement.com
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com