Re: [OSL | CCIE_Voice] UCCX Scripts End Step

[Baktha Muralidharan]

I would think/hope so. I suspect the branches for "successful" and
"timeout", which are mutually exclusive,
are implemented using an "if-then-else" type of construct.

[But then again, no better way to be sure than to lab it out!]

thanks,
/Baktha

--
>
> Message: 1
> Date: Thu, 26 Apr 2012 12:31:33 +0200
> From: Juan Lopez 
> To: Ken Wyan 
> Cc: ccie_voice@onlinestudylist.com, Gurpreet Singh Kukreja
>
> Subject: Re: [OSL | CCIE_Voice] UCCX Scripts End Step
> Message-ID:
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi Ken,
> Gurpreet/Peter are correct - you misunderstand as far as the script
> behavour goes: the script goes to the next step if nothing is specified
> under the 'menu' step's  'substep' (eg: no step defined under the
> 'timeout', 'unsuccessful'...will take it to the next step defined after the
> menu step) Therefore it is best to have a 'Goto' defined below the
> 'connect' step so the script can 'terminate' using the 'terminate' step...
>
> hope this somewhat clear - not easy to explain ... ;)
>
> cheers,
> Juan
>
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[OSL | CCIE_Voice] Re :RSVP and CFB

[Baktha Muralidharan]

Hi Juan

I believe CFB is the only media resource in which the location has to be
configured explicitly.
I haven't tested NOT doing this recently and so, don't have information on
whether the impact
(of NOT explicitly configuring location CFB) is limited to RSVP.

Thanks,
/Baktha


--
>
> Message: 1
> Date: Thu, 26 Apr 2012 20:40:01 +0200
> From: Juan Lopez 
> To: CCIE Study 
> Subject: [OSL | CCIE_Voice] RSVP and CFB
> Message-ID:
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
> all,
> just found out and wonder if someone can confirm: unless you place a CFB in
> a location (instead of being placed in a location by means of the CFB's
> device pool location setting) - RSVP will not be triggered ??
>
> So if I leave the location at the CFB set to  "hub_none" (where it then
> should check the device pool's location setting - correct me if I'm wrong),
> and with the location on the device pool set to HQ, no RSVP will be
> triggered.
> When setting the location on CFB pages to 'HQ' , RSVP will be triggered.
>
> Do you also see this in your labs? thanks for the feedback,
> Juan
> -- next part --
> An HTML attachment was scrubbed...
> URL:
> 
>
>
This is easily done, using voice translation rules (VTRs) on H.323/SIP
gateways.
 - Use "answer-address" on the incoming voip dial-peer to field calls from
x and x333.
- Use VTRs to prefix the CALLED NUMBER with, say, "" if call is from
x and
with "", if call is from x.
- This will enable you to have distinct outgoing dial-peers for the two
calls.
- On the way out, strip out the prefix strings, using VTRs.

Thanks,
/Baktha


> --
>
> Message: 2
> Date: Thu, 26 Apr 2012 15:20:19 -0400
> From: Juan Carlos Anzola 
> To: Online Study 
> Subject: [OSL | CCIE_Voice]  ANI based Call Routing
> Message-ID:
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi Guys,
>
> Consider the following scenario:
>
> I have a CUCM Cluster with the Following Extensions:
>
> 2XXX: Sales
> 3XXX: Engineering
>
> I have a single H.323 or SIP PSTN GW.
>
> I have 2 different ITSP: 10.2.2.2 and 10.3.3.3
>
> Right now, All calls are routing properly throug 10.2.2.2
>
> I want calls from 2XXX to be routed out 10.2.2.2 and calls from 3XXX to be
> routed out 10.3.3.3
>
> The requirement is to do this without modifying anything in CUCM.
>
>
> Thanks in advance,
>
> --
> Juan Carlos Anzola
> -- next part --
> An HTML attachment was scrubbed...
> URL:
> 
>
>
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Re: [OSL | CCIE_Voice] IOS NTP

[Baktha Muralidharan]

1. If you have "ntp server...", then "ntp master", if configured, should be
at a lower stratum than the stratum level of the NTP server. I believe we
don't want BOTH "ntp server" and "ntp master command
2. Do I need the ntp update-calendar command? Seems to be to keep the
hardware clock updated, for battery-powered routers. I wouldn't configure
this.
3. "ntp source loopback 0" only when there is an explicit requirement for
NTP traffic to flow over loopback 0.

thanks,
/Baktha

--
>
> Message: 3
> Date: Wed, 25 Apr 2012 08:29:46 +0200
> From: Maik Stokman 
> To: 
> Subject: [OSL | CCIE_Voice] IOS NTP
> Message-ID: 
> Content-Type: text/plain; charset="utf-8"
>
> Hi,
>
>
>
> For configuring IOS ntp I have 3 questions:
>
>
>
> 1.   Do I need the ntp master command? Which stratum is best practice?
>
> 2.   Do I need the ntp update-calendar command?
>
> 3.   Do I need the configure ?ntp source loopback 0? when site b and c
> must use the HQ loopback interface as NTP server
>
>
>
> At this moment I use the following configuration:
>
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Re: [OSL | CCIE_Voice] Router Switch port configurations

[Baktha Muralidharan]

>
> + for switchport trunk on ESW ports. Never had an issue with it.
>
> A related question. On the 3750, unless explicitly indicated, do you leave
> the native VLAN on the trunk port at default (i.e. "1')?
>
> Thanks,
>
> /Baktha
>

>


> --
>
> Message: 1
> Date: Mon, 23 Apr 2012 16:12:48 +1000
> From: Chris 
> To: ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] Router Switch port configurations
> Message-ID:
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi Bill,
> The spanningtree portfast command does nothing for trunk ports, unless one
> of the newer commands with " trunk" keyword is available. I guess there is
> no harm putting it, but not very cool. I would put this if trunk keyword is
> available.
> Chris
>
> On Mon, Apr 23, 2012 at 2:13 PM,  >wrote:
>
> > Send CCIE_Voice mailing list submissions to
> >ccie_voice@onlinestudylist.com
> >
> > To subscribe or unsubscribe via the World Wide Web, visit
> >http://onlinestudylist.com/mailman/listinfo/ccie_voice
> > or, via email, send a message with subject or body 'help' to
> >ccie_voice-requ...@onlinestudylist.com
> >
> > You can reach the person managing the list at
> >ccie_voice-ow...@onlinestudylist.com
> >
> > When replying, please edit your Subject line so it is more specific
> > than "Re: Contents of CCIE_Voice digest..."
> >
> >
> > Today's Topics:
> >
> >   1. Re: Router Switch port configurations (Bryan Byrne)
> >   2. Re: Router Switch port configurations (Chris)
> >   3. CUE Messaging and call forwarding (Chris)
> >
> >
> > --
> >
> > Message: 1
> > Date: Sun, 22 Apr 2012 12:12:19 -0400
> > From: Bryan Byrne 
> > To: Bill Lake 
> > Cc: Chris ,   "ccie_voice@onlinestudylist.com"
> >
> > Subject: Re: [OSL | CCIE_Voice] Router Switch port configurations
> > Message-ID: 
> > Content-Type: text/plain; charset="iso-8859-1"
> >
> > I agree with Bill. I've had very spotty behavior not including the
> > switchport mode trunk command.
> >
> > -Bryan
> >
> >
> > On Apr 22, 2012, at 10:53 AM, Bill Lake wrote:
> >
> > > I would recommend #2 but you forgot
> > >
> > > switchport voice vlan 11
> > > Spanning-tree portfast (use depends on requirements)
> > >
> > > Reasoning for the preference is that iOS in lab is not totally reliable
> > with #1.  Might work but then it might stop an hour or 10 hours after you
> > leave but that might cause you to have a very expensive lunch
> > >
> > > On Sunday, April 22, 2012, Chris wrote:
> > > Hi All ,
> > >
> > > I don't have a 4-port or 9-port POE switch module to try it on.
> > Therefore I would like for some one to confirm if both or one of
> following
> > port configuration will work on these cards. I do understand the concept,
> > but don't want to find the actual syntax on lab day :). Thanks in
> advance.
> > >
> > > Vlan 10 is DATA
> > > Vlan 11 is VOICE
> > >
> > > Preference 1-
> > > interface FastEthernet0/1/0
> > >  switchport access vlan 10
> > >  switchport voice vlan 11
> > >  spanning-tree portfast
> > > Preference 2-
> > > interface FastEthernet0/1/0
> > > switchport trunk native vlan 10
> > > switchport mode trunk
> > >
> > > Best Regards
> > > Chris
> > > ___
> > > 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
> >
> > -- next part --
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> >
> > --
> >
> > Message: 2
> > Date: Mon, 23 Apr 2012 12:29:03 +1000
> > From: Chris 
> > To: Bryan Byrne 
> > Cc: "ccie_voice@onlinestudylist.com" 
> > Subject: Re: [OSL | CCIE_Voice] Router Switch port configurations
> > Message-ID:
> > e...@mail.gmail.com
> > >
> > Content-Type: text/plain; charset="iso-8859-1"
> >
> > Thanks, Kevin/Bill/Bryan for you suggestions. Much appreciated.
> >
> > Chris
> >
> >
> > On Mon, Apr 23, 2012 at 2:12 AM, Bryan Byrne 
> wrote:
> >
> > > I agree with Bill. I've had very spotty behavior not including the
> > > switchport mode trunk command.
> > >
> > > -Bryan
> > >
> > >
> > > On Apr 22, 2012, at 10:53 AM, Bill Lake wrote:
> > >
> > > I would recommend #2 but you forgot
> > >
> > > switchport voice vlan 11
> > > Spanning-tree portfast (use depends on requirements)
> > >
> > > Reasoning for the preference is that iOS in lab is not totally reliable
> > > with #1.  Might work but then it might stop an hour or 10 hours after
> you
> > > leave but that might cause you to have a very expensive lunch
> > >
> > > On Sunday, April 22, 2012, Chris wrote:
> > >
> > >> Hi All ,
> > >>
> > >> I don't have a 4-port or 9-port POE switch module to try it on.
> > Therefore
> > >> I would like for some one to confirm if both or one of

Re: [OSL | CCIE_Voice] UCCX and Media Resources (Chris)

[Baktha Muralidharan]

Chris

you might try "debug sccp all" (on HQ gateway) to check

 1. Call manager's attempt to allocate a xcoder from HQ
 2. errors, if any, from HQ gateway, as to why the allocation is failing.

thanks,
/Baktha


> --
>
> Message: 1
> Date: Tue, 17 Apr 2012 18:17:24 +1000
> From: Chris 
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] UCCX and Media Resources
> Message-ID:
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
> My UCCX is in HQ device pool. The DP has MRGL allocated to with registered
> transcoder resources. However, when I try to dial from BR1/BR2. The call
> fails to connect. The SDI traces on the call manager show following
> messages:
> *04/17/2012 15:28:30.231
> CCM|MediaManager(9)::disconnOnResourceAllocationFailure, ERROR
>  disconnOnResourceAllocationFailure - fails to allocate
>
> MTP/XCoder,connCount=2|
> *
> Xcoder resource is configured as
> *Transcoding Oper State: ACTIVE - Cause Code: NONE*
> *Active Call Manager: * *10.10.100**.12, Port Number: 2000*
> *TCP Link Status: CONNECTED, Profile Identifier: 1*
> *Reported Max Streams: 6, Reported Max OOS Streams: 0*
> *Supported Codec: g711ulaw, Maximum Packetization Period: 30*
> *Supported Codec: g711alaw, Maximum Packetization Period: 30*
> *Supported Codec: g729ar8, Maximum Packetization Period: 60*
> *Supported Codec: g729abr8, Maximum Packetization Period: 60*
> *Supported Codec: g729r8, Maximum Packetization Period: 60*
> *Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30*
> *Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30*
> *Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization
> Period: 30*
> *
> *
> *MTP Oper State: ACTIVE - Cause Code: NONE*
> *Active Call Manager: * *10.10.100**.12, Port Number: 2000*
> *TCP Link Status: CONNECTED, Profile Identifier: 3*
> *Reported Max Streams: 20, Reported Max OOS Streams: 0*
> *Supported Codec: pass-thru, Maximum Packetization Period: N/A*
> *Supported Codec: g729r8, Maximum Packetization Period: 60*
> *Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30*
> *Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30*
> *Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization
> Period: 30*
> *RSVP : ENABLED*
> MRGL
> [image: Inline image 1]
> Can someone tell me what am I doing wrong.
> Thanks
> Chris
> --
>
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[OSL | CCIE_Voice] accounting for bw for moh for CAC and QoS

[Baktha Muralidharan]

Hi folks

I am wondering the approach taken by other folks on accounting for the
bandwidth consumed by MoH.

I believe that-

- bandwidth used by multicast moh is NOT used for CAC (including RSVP)
purposes
- bandwidth used by unicast moh IS used for CAC
- in either case, priority bandwidth (in LLQ policy) will have to take them
into consideration
- of course, when the multicast moh flows out of gateway flash, neither CAC
nor QOS care, as it doesn't cross the WAN!

As far as accounting for the [additional] bandwidth for moh,  it would, of
course, depend on the codec used for MoH versus codec used by the call. If
same, OR moh is less than voice codec, I guess nothing more needs to be
done.
If the codec used by MoH is different and of higher bandwidth, then we will
have to make sure that the priority bandwidth accounts for MoH. For
example, if calls use g.729, but moh uses g.711, then, in addition to
allowing for n calls, the "priority " command has to also allow
for the additional bandwidth as follows-


 multicast moh --  1 x   +  x g.729 bandwidth
 unicast  moh  -n x 

Is the above correct?

Also, is it recommended that we do the above regardless of  explicit
requirement to account for moh bw.
Finally, as for codec for MoH, it would seem simpler to make the moh codec
SAME as voice-class codec, again, unless otherwise specified.

Any thoughts/comments?


thanks,
/Baktha
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[OSL | CCIE_Voice] CTI RP vs DN

[Baktha Muralidharan]

Hi Ken

In general, I think the CTI RP is a " device" akin to a "phone"  rather
than a DN. You get to associate lines to a CTI RP.
Per UCM SRND, for "redirection" purposes, the recommended mechanism is CTI
RP.

Not sure it answers your question, however.

thanks,
/Baktha


Message: 2
Date: Wed, 11 Apr 2012 22:48:27 +0530
From: Ken Wyan 
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] CTI RP vs DN
Message-ID:
   
Content-Type: text/plain; charset="iso-8859-1"

If we want to configure auto attendant in CUCM-CUC , the procedure is
create a route point , configure an extension to it & under line
configuration set forward all to voicemail.

We can do same thing without creating a CTI Route Point . Just create a
directory number & set forward all to voicemail under DN configuration. It
works

Why does cisco recommend to create a CTI Route Point (it's an unnecessary
step) ?
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[OSL | CCIE_Voice] the directories button display

[Baktha Muralidharan]

Hi Mann

It would seem that once those phone services are deleted, it is impossible
to get them listed in the "default" order.
that would suggest that we implement WITHOUT deleting the phone services,
like, by pointing the url for the phoen that should NOT show the services,
to "external" and setting the external URL to blank.

however, since this causes the Messages/Voicemail feature, adding it back
in, using the getmessages URL brings back voicemail/Messages button
feature, but it is a 2-step access to VM.

Thanks,
/Baktha

--
>
> Message: 3
> Date: Wed, 11 Apr 2012 21:09:42 +0530
> From: Mann Chaddha 
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice]  the directories button display
> Message-ID:
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi All
>
> I see that Cisco Data Dictionary for 7.0 defines the Priority Field as
> follows:
> "Priority from low to high defines where in a list a Service should appear:
> 1 = top of list, 50 (default) = middle, 100 = bottom"
>
> I ran the following commands after deleting the Services from the CUCM
> Admin,
> run sql insert into telecasterservice
>
> (pkid,Name,NameASCII,Description,URLTemplate,tkPhoneService,EnterpriseSubscription,Priority)
> values('d0059763-cdcc-4be7-a2a8-bbd4aac73f63','Missed Calls','Missed
> Calls','Missed Calls','Application:Cisco/MissedCalls',1,'f',*1*)
>
> run sql insert into telecasterservice
>
> (pkid,Name,NameASCII,Description,URLTemplate,tkPhoneService,EnterpriseSubscription,Priority)
> values('a0eed443-c705-4232-86d4-957295dd339c','Placed Calls','Placed
> Calls','Placed Calls','Application:Cisco/PlacedCalls',1,'f'*,2*)
>
> run sql insert into telecasterservice
>
> (pkid,Name,NameASCII,Description,URLTemplate,tkPhoneService,EnterpriseSubscription,Priority)
> values('0061bdd2-26c0-46a4-98a3-48a6878edf53','Received Calls','Received
> Calls','Received Calls','Application:Cisco/ReceivedCalls',1,'f',*3*)
>
> run sql insert into telecasterservice
>
> (pkid,Name,NameASCII,Description,URLTemplate,tkPhoneService,EnterpriseSubscription,Priority)
> values('7eca2cf1-0c8d-4df4-a807-124b18fe89a4','Corporate
> Directory','Corporate Directory','Corporate
> Directory','Application:Cisco/CorporateDirectory',1,'f',*100*)
>
> And here is the ouotput from the select command for the table:
> admin:run sql select * from telecasterservice
> pkid name
> urltemplate  description
> nameascii   tkphoneservice vendor version enterprisesubscription
> enabled priority tkphoneservicecategory
>  ===
>  ===
> === == == === ==
> ===  ==
> d0059763-cdcc-4be7-a2a8-bbd4aac73f63 Missed Calls
> Application:Cisco/MissedCallsMissed CallsMissed
> Calls1 f  t
> 50   0
> a0eed443-c705-4232-86d4-957295dd339c Placed Calls
> Application:Cisco/PlacedCallsPlaced CallsPlaced
> Calls1 f  t
> 50   0
> 0061bdd2-26c0-46a4-98a3-48a6878edf53 Received Calls
> Application:Cisco/ReceivedCalls  Received Calls  Received
> Calls  1 f  t
> 50   0
> 7eca2cf1-0c8d-4df4-a807-124b18fe89a4 Corporate Directory
> Application:Cisco/CorporateDirectory Corporate Directory Corporate
> Directory 1 f  t
> 100  0
> ca69f2e4-d088-47f8-acb2-ceea6722272e Voicemail
> Application:Cisco/Voicemail  Voicemail
> Voicemail   2 t
> t   10
> admin:
>
>
> Still after re-subscribing the services to the phones, Corp Dir always
> shows up at the top.
>
> If anyone has been able to resolve this issue, please comment.
>
>
> Thanks
> Mann
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>
>
> End of CCIE_Voice Digest, Vol 74, Issue 35
> **
>
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Re: [OSL | CCIE_Voice] UCCX failure updating rmcm user on Call manager

[Baktha Muralidharan]

Thanks to all for the tips/suggestions-

Does uccx automatically create the rmcm application user or do we need to
manually create it?

I don't remember having to create it in the past. whenever, the uccx was
partially integrated, I used
to be able to do the following-

 - go into uccx and create/complete telephony-group and rmcm (resources,
skills etc.)
 - on UCM, associate users to rmcm application user
 - configure application/trigger etc.

thanks,
/Baktha
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[OSL | CCIE_Voice] UCCX failure updating rmcm user on Call manager

[Baktha Muralidharan]

Hi folks

I am having trouble integrating UCCX with call manager. I keep getting the
error

  "Failed to update rmcm user on Cisco Unified CM"

when I click on OK on the UCCX Unified CM configuration page.

I am not up on UCCX; any hints on which log file might tell me more about
the issue?

I tried setting the administrator/jtapi/rmjtapi password, to no avail.

thanks in advance,
/Baktha
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Re: [OSL | CCIE_Voice] High Availabilty using route group

[Baktha Muralidharan]

Krishna

Just to confirm, is the following correct?

  - call comes in through a gateway, perhaps from a TDM PSTN?
  - presumably, connection to the backup service provider is through
another CUBE, perhaps, on another router?


Assuming the CUBEs' interface with call manager is SIP,  here is how you
can achieve HA-

   - configure two SIP trunks, one to each CUBE.
   - configure two RGs
   - configure a route list, and add the two RGs to it in the order that
you desire
   - configure the RP, and point to the RL

You can tweek the service parameters to control when attempt to call
through SP1 should time out and move to SP2.

hope this helps,
/Baktha

--
>
> Message: 2
> Date: Sun, 1 Apr 2012 07:17:11 -0700 (PDT)
> From: Krishna 
> To: "ccie_voice@onlinestudylist.com" 
> Subject: Re: [OSL | CCIE_Voice] High Availabilty using route group
> Message-ID:
><1333289831.69846.yahoomail...@web46011.mail.sp1.yahoo.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> All,
>
> I;d like to receive feedback and confirmation about implementing the HA
> system, and this is about concept of design and implementation as well.
> please, advise me on this perspective. Here is the existing call flow :
>
> gateway(site)-->cucm--->Route Group> cubes-->Service provider (sip
> trunk)
>
> But, I want this call flow to be redundant by adding the second service
> provider as well. My question: Is it possible using Route Group feature to
> send only calls to SP2 in case SP 1 down,, if not then how can i achieve HA
> with 2 SP using RG's
>
> gateway(site)-->cucm--->Route Group> cubes-->Service provider 1
> (sip trunk)
> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?
> ? ? ?--->?Service provider 2 (sip trunk)
>
>
>
> Your help is much appreciated.
>
> thank you.
>
> Regards,
>
> Krishna Koilada.
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>
>
> End of CCIE_Voice Digest, Vol 74, Issue 1
> *
>
___
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[OSL | CCIE_Voice] switch-QoS-Quick Question

[Baktha Muralidharan]

Hey Juan

I believe the remark class is there to catch those packets that are tagged
with the EF but are NOT RTP and remark them to scavenger class.
We don't want to remark ALL packets (with DSCP values that are NOT
EF/CS3/AF31), which is what class-default would end up doing.

Similarly for cs3 and af31.

thanks,
/Baktha


On Sat, Mar 31, 2012 at 5:04 PM, wrote:

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> Today's Topics:
>
>   1. Re: switch-QoS-Quick Question (Baktha Muralidharan)
>   2. Re: switch-QoS-Quick Question (George Goglidze)
>   3. Re: switch-QoS-Quick Question (Juan Lopez)
>
>
> ------
>
> Message: 1
> Date: Sat, 31 Mar 2012 13:30:08 -0400
> From: Baktha Muralidharan 
> To: ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] switch-QoS-Quick Question
> Message-ID:
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
> Yes, policing (and marking) is normally done on the ingress interface.
> On the egress side, you can rate-limit, using  a shaper policy.
> The salient difference between policing and shaping is that policer will
> drop or remark, whilst shaper will buffer.
>
> thanks,
> /Baktha
>
> > --
> >
> > Message: 1
> > Date: Sat, 31 Mar 2012 16:37:57 +0200
> > From: Juan Lopez 
> > To: steven moran 
> > Cc: ccie_voice@onlinestudylist.com
> > Subject: Re: [OSL | CCIE_Voice] switch-QoS-Quick Question
> > Message-ID:
> > 5o-wz6t+-zeavcbpui5+4ercqp7zeukftopc...@mail.gmail.com
> > >
> > Content-Type: text/plain; charset="windows-1252"
> >
> > Steven,
> > As far as I know, you cannot do policing on the egress fa0/1 towards the
> > router...
> >
> > Only way to make sure MGCP towards the router is policed and down-marked,
> > is by applying a policer on the ingress access port connecting the
> server I
> > think.
> >
> > Is it a hard requirement to have it done on the fa0/1 uplink to the
> router?
> > Let me know
> >
> > cheers,
> > Juan
> >
> >
> >
> > Op 28 maart 2012 02:11 schreef steven moran  het
> > volgende:
> >
> > > I?m preparing for the exam and  as you are all aware question
> > > interpretation is really important.  Below is a practice question plus
> > > my config on how to approach it.  I would appreciate it if anyone
> > > could comment on my approach to the question and see if the answer
> > > meets the brief.  I considered running auto qos on the phone and
> > > server ports to mark the traffic at source but this seems excessive
> > > for the question.
> > >
> > > Question
> > > On port fa0/1 which is connected to HQ router, guarantee 16k for MGCP
> > > signaling traffic. Excess traffic should be marked to DSCP 8 and then
> > > transmitted.
> > >
> > >
> > >
> > > mls qos
> > > !
> > > mls qos map cos 0 8 16 24 32 46 48 56
> > > !
> > > mls qos map policed-dscp 24 to 8
> > > !
> > > ip access-list extended 100
> > > permit tcp any any eq 2428
> > > permit udp any any eq 2427
> > > !
> > > class-map class-mgcp
> > > match access-group 100
> > > !
> > > policy-map policy-mgcp
> > > class class-mgcp
> > > set dscp cs3
> > > police 16000 8000 exceed-action policed-dscp-transmit
> > > !
> > > interface fa0/1
> > > service input policy-mgcp
> > > ___
> > > 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 <http://www.platinumplacement.com/>
> > >
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> >
> > ---

Re: [OSL | CCIE_Voice] switch-QoS-Quick Question

[Baktha Muralidharan]

Yes, policing (and marking) is normally done on the ingress interface.
On the egress side, you can rate-limit, using  a shaper policy.
The salient difference between policing and shaping is that policer will
drop or remark, whilst shaper will buffer.

thanks,
/Baktha

> --
>
> Message: 1
> Date: Sat, 31 Mar 2012 16:37:57 +0200
> From: Juan Lopez 
> To: steven moran 
> Cc: ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] switch-QoS-Quick Question
> Message-ID:
> >
> Content-Type: text/plain; charset="windows-1252"
>
> Steven,
> As far as I know, you cannot do policing on the egress fa0/1 towards the
> router...
>
> Only way to make sure MGCP towards the router is policed and down-marked,
> is by applying a policer on the ingress access port connecting the server I
> think.
>
> Is it a hard requirement to have it done on the fa0/1 uplink to the router?
> Let me know
>
> cheers,
> Juan
>
>
>
> Op 28 maart 2012 02:11 schreef steven moran  het
> volgende:
>
> > I?m preparing for the exam and  as you are all aware question
> > interpretation is really important.  Below is a practice question plus
> > my config on how to approach it.  I would appreciate it if anyone
> > could comment on my approach to the question and see if the answer
> > meets the brief.  I considered running auto qos on the phone and
> > server ports to mark the traffic at source but this seems excessive
> > for the question.
> >
> > Question
> > On port fa0/1 which is connected to HQ router, guarantee 16k for MGCP
> > signaling traffic. Excess traffic should be marked to DSCP 8 and then
> > transmitted.
> >
> >
> >
> > mls qos
> > !
> > mls qos map cos 0 8 16 24 32 46 48 56
> > !
> > mls qos map policed-dscp 24 to 8
> > !
> > ip access-list extended 100
> > permit tcp any any eq 2428
> > permit udp any any eq 2427
> > !
> > class-map class-mgcp
> > match access-group 100
> > !
> > policy-map policy-mgcp
> > class class-mgcp
> > set dscp cs3
> > police 16000 8000 exceed-action policed-dscp-transmit
> > !
> > interface fa0/1
> > service input policy-mgcp
> > ___
> > 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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>
>
> End of CCIE_Voice Digest, Vol 73, Issue 117
> ***
>
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Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 73, Issue 111

[Baktha Muralidharan]

Steven,

 if you are required explain in terms of Event1/2/3, you want to explicitly
tag them as such.
 For example, annotate the three events, as shown

 Event1:  Backup call agent sending AUEP to gateway, to inquire about an
endpoint

 AUEP trace lines

 Event2:  Gateway returning (200 OK) with valid I: value (connection
identifier)

200 OK lines

 Event3:  Backup call agent asking for additional info (using AUCX) on the
call

AUCX trace lines


Thanks,
/Baktha

--
>
> Message: 3
> Date: Fri, 30 Mar 2012 08:58:30 +1100
> From: steven moran 
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] MGCP failover
> Message-ID:
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
> Could someone please check if I have answered this question correctly
>
> prove when the subscriber fails, the publisher will work.  While the call
> is active shut
> down the CUCM - Subscriber services.  Debug the MGCP message, steps should
> show:
>
> 1. Backup call agent send the message to the gateway to check the status of
> the Endpoint
> 2. Gateway sends the status of the call to the call agent about the status
> of the call.
> 3. Back up the message send from the call agent to the gateway.
> Put all the steps as event 1, event 2, and event 3 in the text file
>
>
> the Sub is 10.10.210.11 and the Pub is 10.10.210.10
>
>
> HQ-RTR(config)#
> Mar 29 21:01:03.621: MGCP Packet sent to 10.10.210.11:2427--->
> NTFY 331366507 *@HQ-RTR MGCP 0.1
> X: 0
> O:
> <---
> Mar 29 21:01:03.625: MGCP Packet received from 10.10.210.11:2427--->
> 200 331366507
> <---
>
> 
> Primary CA Goes Down. AUEP Sent.
> 
> HQ-RTR(config)#
> Mar 29 21:01:11.076: MGCP Packet sent to 10.10.210.11:2427--->
> RSIP 331366508 *@HQ-RTR MGCP 0.1
> RM: graceful
> <---
> 
> *Step 1. RSIP Sent to Secondary CA. Secondary CA Sends ACK.
> *
>
> Mar 29 21:01:11.080: MGCP Packet sent to 10.10.210.10:2427--->
> RSIP 331366510 *@HQ-RTR MGCP 0.1
> RM: restart
> <---
> Mar 29 21:01:11.132: MGCP Packet received from 10.10.210.10:2427--->
> 200 331366510
> <---
> Mar 29 21:01:11.136: MGCP Packet sent to 10.10.210.10:2427--->
> NTFY 331366512 *@HQ-RTR MGCP 0.1
> X: 0
> O:
> <---
> 
> *Step 2.   2nd CA sends AUEP to audit preserved calls*. This is the AUEP
> for S0/DS1-1/6 (only active channel).
> 
>
> Mar 29 21:01:11.208: MGCP Packet received from 10.10.210.10:2427--->
> AUEP 31 S0/DS1-1/1@HQ-RTR MGCP 0.1
> F: X, A, I
> <---
>
> Mar 29 21:01:11.252: MGCP Packet received from 10.10.210.10:2427--->
> AUEP 36 S0/DS1-1/6@HQ-RTR MGCP 0.1
> F: X, A, I
> <---
> Mar 29 21:01:11.256: MGCP Packet sent to 10.10.210.10:2427--->
> 200 36
> I: 2
> X: 6
> L: p:10-20, a:PCMU;PCMA;G.nX64, b:64, e:on, gc:1, s:on, t:10, r:g,
> nt:IN;ATM;LOCAL, v:L;G;D;T;H;ATM;FXR
> L: p:10-220, a:G.729;G.729a;G.729b, b:8, e:on, gc:1, s:on, t:10, r:g,
> nt:IN;ATM;LOCAL, v:L;G;D;T;H;ATM;FXR
> L: p:10-110, a:G.726-16, b:16, e:on, gc:1, s:on, t:10, r:g,
> nt:IN;ATM;LOCAL, v:L;G;D;T;H;ATM;FXR
> L: p:10-70, a:G.726-24, b:24, e:on, gc:1, s:on, t:10, r:g, nt:IN;ATM;LOCAL,
> v:L;G;D;T;H;ATM;FXR
> L: p:10-50, a:G.726-32, b:32, e:on, gc:1, s:on, t:10, r:g, nt:IN;ATM;LOCAL,
> v:L;G;D;T;H;ATM;FXR
> M: sendonly, recvonly, sendrecv, inactive, loopback, conttest, data,
> netwloop, netwtest
> <---
>
> Mar 29 21:01:11.344: MGCP Packet re
> HQ-RTR(config)#ceived from 10.10.210.10:2427--->
> 200 331366512
> <---
> 
> *STEP 3: CA sends AUCX*. connection number corresponds to the one above
> (i=2)
> 
>
> Mar 29 21:01:11.348: MGCP Packet received from 10.10.210.10:2427--->
> AUCX 61 S0/DS1-1/6@HQ-RTR MGCP 0.1
> I: 2
> F: C, M
> <---
> Mar 29 21:01:11.352: MGCP Packet sent to 10.10.210.10:2427--->
> 200 61
> C: D1d330b200F50001
> M: sendrecv
> <---
> **
> CA asks the GW to notify if ICMP Unreachable is recd for RTP Stream.
> **
>
> Mar 29 21:01:11.356: MGCP Packet received from 10.10.210.10:2427--->
> RQNT 62 S0/DS1-1/6@HQ-RTR MGCP 0.1
> X: 6
> R: R/iu, FXR/t38
> Q: process,loop
> <---
> Mar 29 21:01:11.360: MGCP Packet sent to 10.10.210.10:2427--->
> 518 62  Unknown or Unsupported Package
> <---
> Mar 29 21:01:11.576: MGCP Packet sent to 10.10.210.11:2427--->
> RSIP 331366508 *@HQ-RTR MGCP 0.1
> RM: graceful
> <---
> HQ-RTR(config)#
> Mar 29 21:01:26.136: MGCP Packet sent to 10.10.210.10:2427--->
> NTFY 331366513 *@HQ-RTR MGCP 0.1
> X: 0
> O:
> <---
> Mar 29 21:01:26

Re: [OSL | CCIE_Voice] switch-QoS-Quick Question

[Baktha Muralidharan]

Hello,

Isn't it true that the trust stuff applies ONLY to those packets that are
not "caught" by the class (in the qos policy)?
For packets that are processed by the policy-map, you do a "set cs3 (or
whatever)" under the policy.

thanks,
/Baktha


--
>
> Message: 1
> Date: Thu, 29 Mar 2012 08:06:53 +0200
> From: George Goglidze 
> To: Chris 
> Cc: "ccie_voice@onlinestudylist.com" 
> Subject: Re: [OSL | CCIE_Voice] switch-QoS-Quick Question
> Message-ID: <4557c162-94d1-4c7e-963f-1ec20a440...@gmail.com>
> Content-Type: text/plain;   charset=us-ascii
>
> You have to trust DSCP on interface connected to the router. Routers do
> not set cos bits in dot1q header!!!
>
> Same goes for interface connected to CUCM.
>
>
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Re: [OSL | CCIE_Voice] IPexpert lab 10: F/R hub-and-spoke topology vs. RSVP not applied to the GK DP's location

[Baktha Muralidharan]

Hi Juan

As for CAC, guess one would have to consider

 - peak call volume
 - WAN QoS to decide if location-based CAC is [also] needed.

If WAN QoS guarantees enough bandwidth for the anticipated peak volume,
then, not sure we need the local-based CAC [RSVP or otherwise]

As for pass-through, my experience is you only need the codec that will be
used, in this case, the g.729r8. pass-through would be needed if you are
plan to do such things as T.38 fax.

thanks,
/Batkha



> Message: 3
> Date: Wed, 28 Mar 2012 16:33:29 +0200
> From: Juan Lopez 
> To: CCIE Study 
> Subject:
> Message-ID:
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
> In lab 10A, HQ devices are placed in DP_HQ with location HQ. GK/SIP trunk
> are placed in seperate DP, with location
> hub_none and no RSVP applied.
> So when calling from BR1 phone to CME over GK trunk, no CAC is enforced at
> the UCM. But the calls go over the FR link,
> as the FR setup is hub-and-spoke. Considering this topology (versus MPLS) :
> Shouldn't CAC be enforced (RSVP enabled on the hub_none location) to limit
> the amount of calls from BR1 to HQ - by extension to CME? I understand GK
> CAC can be used to contol the amount of calls between the UCM cluster and
> CME - but in this case we need to have CAC on the link BR1<> HQ too.
>
> Also, what is the use of codec pass-through in the RSVP MTP ? Calls use
> g729 over the WAN (normally) - so what scenario exsits to have codec
> pass-through configured with g729 as fallback? why not simply use g729r8 as
> codec, without the passthrough (considering no video is involved here)
>
> thx for sharing thoughts!
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>
> End of CCIE_Voice Digest, Vol 73, Issue 107
> ***
>
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[OSL | CCIE_Voice] TPKT when CUPC (softphone) makes a call..

[Baktha Muralidharan]

Hello folks

In trying to understand the protocols involved. When calling from CUPC
softphone to another instance of CUPC (in deskphone control mode), I see
some messages/protocols that I could use help with-

here is my call flow-

CUPC (softphone)--CUPC (deskphone control)
  on my PC(10.10.0.102)on UCCX (10.10.210.5)

  10.10.210.11 is Call manager sub.

here are the relevant lines from wireshark-

111:26:01.15   10.10.0.102UDP  10.10.210.1146Source
port: 50001  Destination port: sip
211:26:07.44   10.10.0.102SIP/SDP  10.10.210.11   962Request:
INVITE sip:3002@10.10.210.11, with session description
311:26:07.52   10.10.210.11   SIP  10.10.0.102414Status:
100 Trying
411:26:07.53   10.10.210.11   SIP  10.10.0.102832Status:
180 Ringing
511:26:07.53   10.10.210.12   TCP  10.10.0.102   1314[TCP
segment of a reassembled PDU]
611:26:07.53   10.10.210.12   SIP/XML  10.10.0.102949Request:
NOTIFY sip:HQ2@10.10.0.102:50018;transport=TCP
711:26:07.54   10.10.0.102TCP  10.10.210.126650018 >
51919 [ACK] Seq=1 Ack=2132 Win=68 Len=0 TSval=61597619 TSecr=85867605
811:26:07.65   10.10.210.5TPKT 10.10.0.102104
Continuation
911:26:07.65   10.10.210.5TPKT 10.10.0.102 82
Continuation
10   11:26:07.65   10.10.210.5TPKT 10.10.0.102208
Continuation
11   11:26:07.65   10.10.210.5TPKT 10.10.0.102 97
Continuation
12   11:26:07.65   10.10.0.102TCP  10.10.210.5 5449659 >
ms-wbt-server [ACK] Seq=1 Ack=276 Win=269 Len=0
13   11:26:07.73   10.10.0.102SIP  10.10.210.12   539Status:
200 OK
14   11:26:07.80   10.10.210.12   TCP  10.10.0.102 6651919 >
50018 [ACK] Seq=2132 Ack=474 Win=2264 Len=0 TSval=85867875 TSecr=61597638
15   11:26:10.86   10.10.0.102T.12510.10.210.51101003
16   11:26:10.86   10.10.0.102T.12510.10.210.51101003


What are the TPKT packets from UCCX? They source port number for those
packets is 3389. the destination port is an ephemeral port (49659).
Port number 3389 is Microsoft Terminal Server
(RDP)
officially registered as Windows Based Terminal (WBT).

Thanks,
/Baktha
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Re: [OSL | CCIE_Voice] respond 200OK for sip reinvite on CUBE

[Baktha Muralidharan]

Hello,

do you know WHY the re-INVITEs are coming in ? Unless there is a defect of
some kind, every re-INVITE has a purpose in life and so, not sure you want
to block it. Examples of situations triggering re-INVITEs would be -
codec/media change, session expiry timer, supplementary services (Hold,
Transfer).

 - how often are they coming in?
 - what is the interface (voip protocols) with call manager?
 - if you don't block it, what does the CUBE do with the re-INVITE?

thanks,
/Baktha

>
>Hi All,
>
>I did my best in resolving the sip reinvite issue for AS5400XM cube, but
couldn't find the solution for Version 12.4.24 -T5. The issue is the
carrier sending the sip reinvite intermittently even min-se >is set to
1800(30 minutes) for the established call, and I tried to block that sip
reinvite at CUBE level, and to respond with 200ok to keep the session
alive, but due to no response from the CUBE, >the session is being
terminated by the vendor/carrier by sending the bye . I am figuring out a
way to fix this problem rather updating the IOS to 15 version, and
implementing the "mid-call signalling >?block" under voice services. Any
help on this greatly appreciated and admirable.
>
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[OSL | CCIE_Voice] CUPC Signalling (Ken Wyan)

[Baktha Muralidharan]

So, would https be considered a "signaling" protocol at all?
what about for IPPM ?

sorry not clear how the protocol layering works when https is involved.

thanks,
/Baktha
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[OSL | CCIE_Voice] QoS config

[Baktha Muralidharan]

Hi folks

when asked to configure WAN QoS between two sites, with just the following
specifics-

  - allow 2 calls
  - RTP compression on the link

do you do the whole SRND-recommended, 2-level qos policy, including 95% bw
calculation ?
or do you just do auto qos (trust, if we can assume packets are marked
appropriately) and just make sure that the map-class for the other site is
adjusted to allow sufficient bandwidth for calls?

I tend to do the SRND-compliant config, as it allows us to do compression
and shaping at class level, thereby eliminating the need to configure
map-class for the other link (which would otherwise end up with the
default bandwidth of 56kbps)

I am interested in learning what could be gotchas or which words/terms to
particular attention to, in deciding whether to go with the hierarchical
policy versus sticking with what's generated by auto qos.

thanks in advance,
/Baktha
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Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 73, Issue 78

[Baktha Muralidharan]

hey Juan

it would seem like you are running into the following defect-


CSCsw73779 - Direct Calls to Unity Using AAR Route Incorrectly
*CSCsw73779 
*
*Direct Calls to Unity Using AAR Route Incorrectly*
** 
**

Issue:

Two site centralized CCM deployment and Unity is on Site A.
Customer is setting up AAR between the sites and is in the process to
configure AAR for unity.
Everything works fine however if a phone user presses the voicemail
button (direct call to voicemail and there is not enough bandwidth)
the call routes to the PSTN.
This is all fine except that setup contains forwarding/redirecting
info as follows:


*CCM|Ie - Ni2BearerCapabilityIe IEData= 04 03 80 90 A2
*CCM|Ie - PriDmsChannelIdIe IEData= 18 03 A9 83 97
*CCM|Ie - Q931DisplayIe IEData= 28 1E B1 54 6F 20 6F 75 74 73 69 64 65
20 2D 20 70 68 6F 6E 65 20 34 31 36 33 34 38 30 39 39 31
*CCM|Ie - Q931CallingPartyIe IEData= 6C 0C 21 81 34 31 36 33 34 38 30 39 39 31
*CCM|Ie - Q931CalledPartyIe IEData= 70 0B A1 34 31 36 39 37 34 32 36
30 3 
[Edit this 
enclosure]
*Release-note: Added 12/22/2008 15:48:28 by
deepagup
*
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*Release-note: Added 12/22/2008 15:48:28 by
deepagup
*

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*Email Discussion: Added 12/22/2008 15:47:25 by
deepagup
*



[Wrap 
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*Email Discussion: Added 12/22/2008 15:47:25 by
deepagup
*
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*Email Discussion: Added 12/22/2008 15:47:25 by
deepagup
*
**Don't recall running into this. what version of CUCM are you running?

Could you add a VTR on the gateway to zap out RDNIS if it is equal to
voicemail pilot ?

thanks,
/Baktha




--
>
> Message: 3
> Date: Mon, 19 Mar 2012 21:58:21 +0100
> From: Juan Lopez 
> To: CCIE Study 
> Subject: [OSL | CCIE_Voice] CUC and AARneed some help
> Message-ID:
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
> All, I need some help:
>
>  I have CUC setup at the HQ. When using AAR between SB and HQ, when the SB
> phones presses the 'voicemail' button, I would have thought this
> constitutes a direct call (aka: no RDNIS sent) to voicemail. However, I
> notice the call is being sent out with the RDNIS of the voicemail hunt
> pilot. So the caller is connected to the welcome prompt instead...
>
> this really strikes me - I did not see that one coming... Is this normal?
> What can be done so that a direct call to VM from a SB phone - to listen to
> his/her voicemail messages - is sent to the subscribers' inbox instead? (I
> cannot uncheck de delivery of RDNIS at the SB GW, as this needs to be sent
> for forwarded calls)
>
> any help is greatly appreciated.
> Juan
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>
>
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Re: [OSL | CCIE_Voice] sip & sipv2

[Baktha Muralidharan]

Perhaps, you are referring to "session protocol sipv2" in dial-peers ?
I don't recall  sesing a "session protocol sip" or "session protocol sipv1".
Don't think you need to worry about it :-)

thanks,
/Baktha


> --
>
> Message: 1
> Date: Sun, 18 Mar 2012 08:26:48 -0700 (PDT)
> From: "eng_firasoq...@yahoo.com" 
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] sip & sipv2
> Message-ID:
><1332084408.48418.yahoomailclas...@web161805.mail.bf1.yahoo.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Dear all,
> ?
> ?Can you tell me what is the difference between sip & sip v2
> ?
> Regards
> ?
> ?
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Re: [OSL | CCIE_Voice] Hardware VPN challenges

[Baktha Muralidharan]

Hello,

I have been having issues connecting via hardware VPN too. I tried removing
the FW-IN ACL from the outside interface, to no avail.
here is what I see in the logs-

000287: *Mar 21 22:08:09: ISAKMP:(0):Checking ISAKMP transform 1 against
priority 65527 policy
000288: *Mar 21 22:08:09: ISAKMP:  encryption 3DES-CBC
000289: *Mar 21 22:08:09: ISAKMP:  hash SHA
000290: *Mar 21 22:08:09: ISAKMP:  default group 2
000291: *Mar 21 22:08:09: ISAKMP:  auth XAUTHInitPreShared
000292: *Mar 21 22:08:09: ISAKMP:  life type in seconds
000293: *Mar 21 22:08:09: ISAKMP:  life duration (VPI) of  0x0 0x20
0xC4 0x9B
000294: *Mar 21 22:08:09: ISAKMP:(0):atts are acceptable. Next payload is 0
000295: *Mar 21 22:08:09: ISAKMP:(0):Acceptable atts:actual life: 2147483
000296: *Mar 21 22:08:09: ISAKMP:(0):Acceptable atts:life: 0
000297: *Mar 21 22:08:09: ISAKMP:(0):Fill atts in sa vpi_length:4
000298: *Mar 21 22:08:09: ISAKMP:(0):Fill atts in sa life_in_seconds:2147483
000299: *Mar 21 22:08:09: ISAKMP:(0):Returning Actual lifetime: 2147483
000300: *Mar 21 22:08:09: ISAKMP:(0)::Started lifetime timer: 2147483.

000301: *Mar 21 22:08:09: ISAKMP (0:0): vendor ID is NAT-T RFC 3947
000302: *Mar 21 22:08:09: ISAKMP:(0): processing KE payload. message ID = 0
000303: *Mar 21 22:08:10: ISAKMP:(0): processing NONCE payload. message ID
= 0
000304: *Mar 21 22:08:10: ISAKMP:(1004): processing HASH payload. message
ID = 0
000305: *Mar 21 22:08:10: ISAKMP:(1004): Hash payload is incorrect!
<<
000306: *Mar 21 22:08:10: ISAKMP (0:1004): Unknown Input
IKE_MESG_FROM_PEER, IKE_AM_EXCH:  state = IKE_I_AM1
000307: *Mar 21 22:08:10: ISAKMP:(1004):Input = IKE_MESG_FROM_PEER,
IKE_AM_EXCH
000308: *Mar 21 22:08:10: ISAKMP:(1004):Old State = IKE_I_AM1  New State =
IKE_I_AM1
 <
000309: *Mar 21 22:08:10: %CRYPTO-6-IKMP_MODE_FAILURE: Processing of
Aggressive mode failed with peer at 74.126.20.247 <<
000310: *Mar 21 22:08:19: ISAKMP:(1004): retransmitting phase 1
AG_INIT_EXCH...


Not sure if the two lines

"Hash payload is incorrect.." and
"Processing of Aggressive mode failed with peer..."


suggest anything.

Could you help?

Thanks in advance,
/Baktha
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Re: [OSL | CCIE_Voice] [OSL | CCIE_VOICE] SRST and MWI unsolicited

[Baktha Muralidharan]

Hi

hope your issue is not with ephone-dn 1. it is missing "mwi sip"

> ephone-dn? 1
> number 4001
> description +1442077964001
> name +1442077964001
> call-forward busy 4600
> call-forward noan 4600 timeout 10

"trace ccn subs sip" from CUE will show if a NOTIFY goes out. Use "mwi
refresh all" on CUE to force an MWI notification.

thanks,
/Baktha
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Re: [OSL | CCIE_Voice] Call Routing Strategy

[Baktha Muralidharan]

Folks

"Strategy" is probably too broad a term, but regardless of the specifics, I
use the following "strategy"

1. reading/interpreting-
 - pay attention to DNIS ton
 - pay attention to ANI plan/ton
 - pay attention to display requirements, especially when there is a
backup route requirement
 - pay attention to how you reference the RG, using LRG?
 - don't overconfigure.

2. timing/order

 - I think it is recommended that SRST testing be done as early as
possible
 - it would then follow that call routing be done and tested early as
well

3. config

- H.323/SIP ? do the DNIS/ANI manipulations on router
- backup route?  do all the manipulations on RG
- ("To:") display requirements? manipulate DNS on RP, just for display
purposes
- CFUR calls from different dial domains? use called party
transformation CSS
- unless required otherwise, use E.164 for CFUR and AAR

thanks,
/Baktha


Today's Topics:
>
>   1. Re: Call Routing Strategy (Steve Sarrick)
>   2. Re: Call Routing Strategy (Juan Carlos Anzola)
>
>
> --
>
> Message: 1
> Date: Fri, 16 Mar 2012 09:59:03 -0400
> From: Steve Sarrick 
> To: Jason Murray , 
> Subject: Re: [OSL | CCIE_Voice] Call Routing Strategy
> Message-ID: 
> Content-Type: text/plain; charset="iso-8859-1"
>
> I thought the same at one point but now I have adopted more of a ?read it
> and do what it says, nothing more? strategy.  You can get caught up with
> this mentality and then adjusting to a question that may appear that
> eliminates the need to use translation/transformation takes more time.  I
> create my partitions/css and apply to DP/Gateways to be ready for
> transforming but don?t apply unless my iniital read takes me down that
> path.
> More and more of the practice labs I see have some type of hybrid of
> straight RP versus Globalized/Localized, plus dialing, etc.
>
>
> On 3/16/12 9:29 AM, "Jason Murray"  wrote:
>
> > So I am trying to get a good strategy and stick with it as far as call
> routing
> > goes.  I wanted to get your opinions on how you are tackling the
> strategy for
> > call routing.  I am finding that it seems to do routing as follows covers
> > pretty much all bases as well as being more flexible : Using Translation
> > Patterns that change the digits to +E164 then it points to a Route
> Pattern
> > that points to a Gateway that does ANI and DNIS transformations covers
> > everything.   But as I look at some labs in vol 2 it seems to be a
> little over
> > kill sometimes and just doing digit manipulations in the Route Pattern or
> > Route list would suffice.  I want to get a good strategy down and stick
> with
> > it for going into the lab so no matter what they throw at me I can be
> covered.
> >
> >
> >  Whats everyones take on this, what are you using as a strategy?  Are you
> > going to just go into the lab and look at the routing first to determine
> what
> > strategy to use that will take up less time?
> >
> >
> >  Thanks
> >  Jason
> >
> >
> > ___
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> > visit www.ipexpert.com
> >
> > Are you a CCNP or CCIE and looking for a job? Check out
> > www.PlatinumPlacement.com
>
>
>
> Steve Sarrick
> SE - PA Select
> 412-480-3861
>
>
> Join our LinkedIN Group for PA Select opportunities at:
> http://www.linkedin.com/groups?about=&gid=4133756
>
>
>
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> --
>
> Message: 2
> Date: Fri, 16 Mar 2012 11:47:59 -0400
> From: Juan Carlos Anzola 
> To: Steve Sarrick 
> Cc: ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] Call Routing Strategy
> Message-ID:
> >
> Content-Type: text/plain; charset="windows-1252"
>
> Hi Jason/Steve,
>
>Great post, Yesterday i was designing a "Standard DialPlan" that could
> adapt to any(hopefully) requirement and have the same question, is that a
> good strattegy?
>
>Since i am going for my first attempt i can't tell, but for me it was a
> great excercise to design on paper this DialPlan considering different
> crazy requirement that challenges its adaptability.
>
>   Has someone already tried to do this? How was the outcome?
>
>
> Regards,
>
>
>
>
>
>
> On Fri, Mar 16, 2012 at 9:59 AM, Steve Sarrick  wrote:
>
> >  I thought the same at one point but now I have adopted more of a ?read
> > it and do what it says, nothing more? strategy.  You can get caught up
> with
> > this mentality and then adjusting to a question that may appear that
> > eliminates the need to use translation/transformation takes more time.  I
> > create my partitions/css and apply to DP/Gateways to be ready for
> > transforming but

Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 73, Issue 57

[Baktha Muralidharan]

Couldn't agree with Emanuel more!  You really need to figure out your own
strategy!

On a slightly different note, I would like to add this-

This is a great forum for learning. PLEASE, let us not undermine the value
of this with posts that either directly discuss questions from the lab or
otherwise violate NDA. I know, it can be extremely frustrating when your
nth attempt was  unsuccessful, and may feel like blowing off with
specifics/details of what was asked and how you did what you did and still
failed etc. etc. [Trust me, I am speaking from experience- I haven't passed
yet!]. But, given this is a public forum, [and accessible to Cisco]  if we
get too indiscreet, we could well hurt ourselves, by causing them to change
the exam [again]!  and that doesn't help any of us, especially those who
are planning on going back!


All the best for your exam Emanuel!
/Baktha

--
>
> Message: 2
> Date: Wed, 14 Mar 2012 10:00:18 -0300
> From: Emanuel Damasceno 
> To: ccie v2012 
> Cc: ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] CCIE Voice tutoring
> Message-ID:
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hey man,
>
> I think from my experience you need to start analyzing the exams. We all
> know how to answer the exam correctly here, if we have a reasonable amount
> of time. But as you solve the same lab over and over again you start
> realizing you can do things faster and in a different form. Everybody has
> their own strategy. I spoke with almost ALL CCIE Voice here in Brazil, and
> some of them even see me writing this. I heard about doing the exam
> question by question, I heard about doing first CUCM and softwares then
> CLI, and I also heard about CLI then CUCM and softwares.
>
> You need to take notes as you go. You need to consult who has been there.
> None of them will disclose Cisco's NDA, but they will tell you if you are
> in the right track. I heard many things about my strategy, constructive and
> destructive. Some people don't agree with my strategy, but that's the way
> it works for me.
>
> Try to be analytical. Solve one question and start researching the commands
> you used. Use the command reference guide, use SRND, watch the videos from
> IPE and INE closely. They do give you hints on what to do or what you might
> see in the exam. You need to think outside the box, you need to research
> about problems that will make your phones never register, and such.
>
> My strategy is explained on YouTube by Matthew Berry. All you need to do is
> type "Device Based Approach". I think he makes a lot of sense, but some
> people don't agree with it. You need to see other strategies so you can
> perfect yours.
>
> Everybody here knows how to configure QoS, Media Resources, MOH, MGCP, SIP,
> H323 But what we need to do is analyze on how to do it with precision
> and speed. It doesn't matter if you have speed but you have to go back to
> your devices over and over again.
>
> My exam is in 9 days. Right now I left it in the hands of God. Come what
> may, if I don't get it this time, God and my friends know I gave my best.
> It was just not the time yet.
>
> I hope this helps.
> Best regards
> *Emanuel Damasceno*
> CCNP Voice
>
>
>
>
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Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 73, Issue 19

[Baktha Muralidharan]

Hi Mathew


are you trying to call another phone in the same site (as the router
hosting the MVA service) or another site?
could you post the dial-peer you are using to point to the MVA pilot?

assuming that the extn you are calling is either in the None partition or
is callable from the gateway hosting the MVA service.

thanks,
/Baktha


--
>
> Message: 2
> Date: Tue, 06 Mar 2012 10:29:55 -0600
> From: Mathew Miller 
> To: 
> Subject: [OSL | CCIE_Voice] Mobile Voice Access
> Message-ID: 
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hello All,
>
> I have setup mobile voice access for use to be able to dial extensions
> using
> enterprise access in my home lab. 100 times out of 100 it works just
> peachy?
>
> I setup my dial-peer on my router, I setup the IVR service on the router, I
> setup my RDP and RD, along with my Mobile Voice Access number and set my
> Service Parameters.
>
> In my lunch dates I set this up exactly like I have done 100 times at home
> and everything seems to work fine until I try to dial the the extension I
> want to call and I get the Annunciator telling me my call can't be
> completed
> as dialed.
>
> I have checked my re-routing CSS and all the steps in setup and have access
> to internal extensions so I don't know what I am doing wrong. I have tried
> to create the issue in my home lab and cant seem to do it. It work EVERY
> time in my home lab.
>
> Can anyone think of something I may be overlooking?
>
>
>
>
>
>
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>
>
> End of CCIE_Voice Digest, Vol 73, Issue 19
> **
>
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[OSL | CCIE_Voice] Answer too soon timer on Remote Destination (for SNR)

[Baktha Muralidharan]

Just a confirmation request--

Is it true that the "Answer too soon timer" on Remote Destination config
starts *AFTER* "Delay Before Ringing timer".
It seems to, but wanted to be sure. Help page is not very clear about it.

thanks in advance,
/Baktha
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[OSL | CCIE_Voice] Voiceview express

[Baktha Muralidharan]

Hi folks

I configured voiceview express on CME (Branch2 router). here is my config
on CME-

   url services http://http://%3ccue/> IP>/voiceview/common/login.do
   url authentication http://http://%3ccme/>IP>/CCMCIP/authenticate.asp
   authentication credential username pwd (this is same as what's
configured under CUE GUI-->CallManager page)

On CUE, voiceview is configured/enabled.

On the phone, if I hit the Services button, I see "CME Services URL". If I
choose it, it is stuck in "Requesting..." state

On CME, debug ip http all shows the  folowing-

 HTTP GET comes in..

  then the following-

* service_url_main_page: CP send failed error=4748*

Any hints on what might be wrong?

   I have tried reloading the router, to no avail.

thanks in advance,
/Baktha
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Re: [OSL | CCIE_Voice] CME MWI Error (using SIP MWI)

[Baktha Muralidharan]

Ken,

can you explain why you need mwi-relay?

Thanks,
/Baktha

--
>
> Message: 1
> Date: Wed, 29 Feb 2012 10:37:48 +0530
> From: Ken Wyan 
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] CME MWI Error (using SIP MWI)
> Message-ID:
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
> When I configure mwi relay  command under telephony-service ,  I get
> following error message
>
> "Error registring to service SIP_MWI, reason MWI_API_CLIENT_EXISTS"
>
> What may be the reason?
>
> Ken
> -- next part --
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Re: [OSL | CCIE_Voice] MRGL order

[Baktha Muralidharan]

Hi AJ

First, as you know, when in doubt, place each  media resource in a separate
MRG.. to play it safe.

I believe that the only two media resources that might be interchangeably
used (and hence interchangeably selected) are MTP and transcoder.
Consequently, the order of [MRGs with] the others, namely, MoH, CFB,
Annunciator, and RSVP MTPs do not matter.

Since transcoders take up DSP resources, you want to make sure you put it
in a separate MRG (i.e. different from the MRG containing MTPs) and place
it BELOW the MRG with MTP.
Finally, the following are the considerations for choosing (and hence
ordering) call manager MTPs versus IOS MTPs

  - capabilities
  - capacity (UCM MTPs are around 48 or something like that; you can have
upto 500 IOS sw MTPs)
  - location of router and UCM and whether you want your call to make that
extra round trip to pick up an MTP at another location

thanks,
/Baktha

Message: 2
Date: Mon, 27 Feb 2012 21:18:25 -0800
From: AJ BG 
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] MRGL order
Message-ID:
   
Content-Type: text/plain; charset="windows-1252"

Hello All,

If you have multiple resources with overlapping functionality then what is
the correct MRG order?
Let say there are   MTP hardware, Server MTP, Transcoding, and RSVP
I would configure it this way
MRGL-HQ
1. MRG-MTP-Hardwar
2. MRG-MTP-Servers
3. MRG-Xcoding-HQ
4. MRG-RSVP-HQ
Is this the correct order? I don?t want my transcoding or RSVP used as MTP.
If you need to invoke a  MTP resources, but there is no preference in the
workbook, is it ok to use the servers? MTP? Or should we create a MTP
resource in the router?
Thanks,
AJ
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Re: [OSL | CCIE_Voice] AAR with RSVP

[Baktha Muralidharan]

Hi Rynard,

Assuming it is a h.323 gateway, please look into "debug h225 asn1".
Not seeing a call in ISDN debug may not imply the call didn't hit the
gateway.
Also, IIRC, reorder tone means the call has hit the gateway and annunciator
means the call manager doesn't know how to route the calls.

thanks,
/Baktha

Message: 3
Date: Tue, 28 Feb 2012 06:58:58 +
From: Rynard Coetzee 
To: "ccie_voice@onlinestudylist.com" 
Subject: Re: [OSL | CCIE_Voice] AAR with RSVP
Message-ID:
   <
97D19F256B859A48A6A93E8A9210E52A17AF8FCE@BYTESEXCH2K10N1.bytes.local>
Content-Type: text/plain; charset="us-ascii"

Hi Bakta
External number mask on the line is full E.164 number (+442077964001) I
have no AAR prefix configured The route pattern I have is
\+.442077964001and this RP is in a separate PT  ,strange thing is I
don`t see the call
getting to the GW when I do Q931 debug ,my RP is pointing to the Local RG.
Regards
Rynard
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Re: [OSL | CCIE_Voice] AAR with RSVP

[Baktha Muralidharan]

Hi Rynard


could you describe the following-

 - External number mask on the line
 - AAR prefix
 - route pattern you have to route the AAR call

thanks,
/Baktha



>
> Today's Topics:
>
>   1. AAR with RSVP (Rynard Coetzee)
>
>
> --
>
> Message: 1
> Date: Mon, 27 Feb 2012 11:40:53 +
> From: Rynard Coetzee 
> To: "ccie_voice@onlinestudylist.com" 
> Subject: [OSL | CCIE_Voice] AAR with RSVP
> Message-ID:
>
>  <97D19F256B859A48A6A93E8A9210E52A17AF8F5A@BYTESEXCH2K10N1.bytes.local>
> Content-Type: text/plain; charset="us-ascii"
>
> Hi All
> I have a issue where my AAR is not working from HQ site to BR2 site ,I
> have configured the RSVP agent and tested it ,working 100% ,I also have AAR
> group ,and AAR RP configured and also assigned AAR CSS to phones ,the group
> is also configured under the line. I have also enabled AAR under service
> params ,when I disable the RSVP agent to force the call via the PSTN ,I see
> the "Network Congestion, Rerouting" text on the phone ,but I get a fastbusy
> ,and also running debugs on the HQ GW ,I don`t see the call hitting the GW.
> Not sure what I am missing here ,I also tried disabling the RSVP
> completely and using Location B/Width at 23K ,but same result ?
> Any ideas ?
> Regards
> Rynard
>
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> CCIE_Voice mailing list
> CCIE_Voice@onlinestudylist.com
> http://onlinestudylist.com/mailman/listinfo/ccie_voice
>
>
> End of CCIE_Voice Digest, Vol 72, Issue 144
> ***
>
___
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Re: [OSL | CCIE_Voice] UCCX & Presen

[Baktha Muralidharan]

Hi Abel


I am sorry you lost points in UCCX/Presence section.
As for UCCX section, given that you have verifed everything, it is hard to
think of what might have gone wrong-

- were you required to allow other sites to call UCCX? If yes, might need a
transcoder at the UCCX site and need to set the device pool (in UCCX)
telephony-group
- unfortunately, there is always the possibility that you mis-interpreted
the requirements or mis-read the details (number of ports, pilot number
etc.)

regards,
/Baktha




> Message: 3
> Date: Sat, 25 Feb 2012 20:27:34 -0400
> From: "Abel ..." 
> To: "eng_firasoq...@yahoo.com" 
> Cc: ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] UCCX & Presence
> Message-ID:
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
> You lost more than 20% on previous sections?
>
> On Sat, Feb 25, 2012 at 11:23 AM, eng_firasoq...@yahoo.com <
> eng_firasoq...@yahoo.com> wrote:
>
> > Hi,
> >
> >  Dears plz if u suppose this NDA violation plz discard this mail.
> >
> > I take my first attempt & I failed.
> >
> >   I get 0 points in UCCX section & presence I dont know why?
> > the uccx script was working fine with me & I tested it 3 times & related
> > to presence it was easy question & I solved it and it was working fine.
> > so my question if the UCCX script was working fine & the presence also
> was
> > working fine what cisco exactly expected from our side, I don't know if
> > they are looking for some thing else so plz can u suggest any thing
> related
> > to uccx & presence so we can achieve cisco requirements in the next
> attempt
> >
> > Regards
> >
> >
> > ___
> > 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
> >
> -- next part --
> An HTML attachment was scrubbed...
> URL:
> 
>
>
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Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 72, Issue 136

[Baktha Muralidharan]

Randall

sorry, not sure what you mean by MGCP TS?
are you referring to "Telnet Session" into an MGCP gateway? or TS =>
Troubleshooting?

If troubleshooting, you want to turn ON

 - debug isdn q921
 - debug mgcp packet

additionally, may want to issue "show voic call status" when the call is
established, to double-check channel used.

not sure I answered your question.

thanks,
/Baktha



> --
>
> Message: 1
> Date: Sat, 25 Feb 2012 10:42:59 -0800 (PST)
> From: Randall Crumm 
> To: Online Study 
> Subject: [OSL | CCIE_Voice] MGCP TS Question
> Message-ID:
><1330195379.19265.yahoomail...@web164601.mail.gq1.yahoo.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> HI,
> I am working on MGCP TS.?
>
> What is the best way to capture the output in the putty session or console?
> ?
> Cheers,
> Randall
>
>
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Re: [OSL | CCIE_Voice] Tragic news (Emanuel Damasceno)

[Baktha Muralidharan]

>
> Very sad indeed. Condolences to his family.


/Baktha



> Message: 1
> Date: Sat, 25 Feb 2012 01:59:03 -0200
> From: Emanuel Damasceno 
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] Tragic news
> Message-ID:
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hello brothers,
>
> I just like to let you know that my friend and study partner, Jeferson
> Guardia CCIE #28157, has passed away yesterday.
>
> He was a skateboarder and day before yesterday, he fell, hitting his head.
> He didn't want to go to the hospital, because he thought he was ok.
> According to his father he passed in his sleep due to a blood clot in his
> brain. This is a tragic moment for all his family and friends.
>
> I thought I should share this with you guys because he's been very active
> here on the list, and we were studying together for the CCIE Voice. He was
> a great motivator and helped me get out of my personal problems so I could
> focus on my studies. It's sad how life is, and what shocks everybody the
> most is that he was only 24 years old (soon to be 25 on March 20th).
>
> Mourning, but still on the fight... =(
>
> *Emanuel Damasceno*
> CCNP Voice
___
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Re: [OSL | CCIE_Voice] cBarge in CME SRST

[Baktha Muralidharan]

Could you pls post the entire config ?

--
>
> Message: 2
> Date: Fri, 24 Feb 2012 05:09:47 +
> From: Rynard Coetzee 
> To: "wormh...@sch.hu" 
> Cc: "ccie_voice@onlinestudylist.com" 
> Subject: Re: [OSL | CCIE_Voice] cBarge in CME SRST
> Message-ID:
>
>  <97D19F256B859A48A6A93E8A9210E52A17AF8D1D@BYTESEXCH2K10N1.bytes.local>
> Content-Type: text/plain; charset="iso-8859-1"
>
> I have tried all and none ,same outcome.
>
> -Original Message-
> From: "Farkas P?ter" [mailto:wormh...@sch.bme.hu]
> Sent: 23 February 2012 03:52 PM
> To: Rynard Coetzee
> Cc: ccie_voice@onlinestudylist.com
> Subject: Re: RE: [OSL | CCIE_Voice] cBarge in CME SRST
>
> What srst provision in your scenario: none/dn/all?
>
> - Original Message -
> From: Rynard Coetzee 
> Date: Thursday, February 23, 2012 7:00 am
> Subject: RE: [OSL | CCIE_Voice] cBarge in CME SRST
> To: "wormh...@sch.hu" 
> Cc: "ccie_voice@onlinestudylist.com" 
>
>
> > Yes I have tried it under the template and under the ephone.
> >
> >  -Original Message-
> >  From: "Farkas P?ter" [
> >  Sent: 22 February 2012 02:58 PM
> >  To: Rynard Coetzee
> >  Cc: ccie_voice@onlinestudylist.com
> >  Subject: Re: [OSL | CCIE_Voice] cBarge in CME SRST
> >
> >  Have you tried to turn off privacy and enable remote-in-use sofktkey
> > through an ephone-template attached to the ephone? Privacy setting on
> ephone has a bug in SRST mode.
> >
> >  Peter
> >  - Original Message -
> >  From: Rynard Coetzee 
> >  Date: Wednesday, February 22, 2012 1:51 pm
> >  Subject: [OSL | CCIE_Voice] cBarge in CME SRST
> >  To: "ccie_voice@onlinestudylist.com" 
> >
> >
> >  > Hi All
> >  >  I have an issue to get the cBarge to work when my H323 GW goes
> > into  > SRST ,the shared line shows up on both phones ,but when I have
> > an  > active call on one phone ,I don`t see the number on the other
> > phone  > ,and the other phone does not go into remote in use state
> > when I press  > the shared line button. I have privacy turned off
> > under the ephones  > and also under the telephony service. Also my CFB
> > is registered to the router when in srst mode ,I am able to make a
> normal ad-hoc conference when in srst mode.
> >  >  Any ideas ?
> >  >  Regards
> >  >  Rynard
> >  > ___
> >  >  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
>
>
> --
>
>
Could you please explain the call flow again? Why are you making call from
gateway to gateway?
are both sites in SRST ?
do you see the call leaving the originating site in "debug isdn q931" ?
Please post the part of "show run" that shows translation rules and
dial-peers.


Message: 3
> Date: Fri, 24 Feb 2012 01:32:18 -0500
> From: 
> To: , 
> Subject: [OSL | CCIE_Voice] [SRST Calls Failing using Multiple
>Gateways for SRST] Any imput appreciated.
> Message-ID:
>
>  <426D14439C8C604B90E332DD4696917301C9706689@SP049EXC32.compucom.local>
> Content-Type: text/plain; charset="us-ascii"
>
> Any help appreciated-Thanks.  Sorry for the long output but really need
> some help with this-->
> SRST Calls Fails using two voice gateways -->
> Calls only fail when going from gateway A to gateway B-->
> Otherwise calls complete fine -->
> Half the ephones register to gateway A -->
> Half the ephones register to gateway B -->
> Calls are received inbound on gateway A -->
> Calls are forwarded to Gateway B via voip dialpeer successfully -->
> When a call is placed to an ephone registered to gateway B call fails -->
> Call is forward via voip dial peer to gateway B successfully  -->
> The call matches inbound dial-peer 1 successfully.  -->
> The called number has a virtual dial peer created  -->
> Inbound call to Router A   -->
> -->to gateway B via voip dial peer to gateway B -->
> -->call fails with recorded messeage  -->
> -->see-->Result=Success(0); Incoming Dial-peer=1 Is Matched in ccapi
> -->later see -->Result=NO_MATCH(-1) After All Match Rules Attempted in
> ccapi
>
> I believe the problem is that I'm not matching an outbound dialpeer, but
> could use some help here.
>
>
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Re: [OSL | CCIE_Voice] LAN QoS, cBarge in CME SRST

[Baktha Muralidharan]

If nothing specific is required for traffic from/to the phones, I would not
do  anything there.
Run the auto qos on an unused port.

Thanks,
/Baktha


> --
>
> Message: 1
> Date: Wed, 22 Feb 2012 02:13:29 -0800 (PST)
> From: Vega Wong 
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] LAN QoS
> Message-ID:
><1329905609.25199.yahoomailclas...@web65905.mail.ac4.yahoo.com>
> Content-Type: text/plain; charset="utf-8"
>
> Hi all
>
> Just want to get some advise regarding LAN QoS.
>
> Let say if the question ask to apply priority queue and specify certain
> drop % based on a COS value to the interface that connect to the router.
> What would you do with the interface that connecting to the phone?
>
> The reason I asked is because if auto-qos is used on the phone ports, a
> few config line will be generated, such as policy map etc. Do you normally
> leave those in there on the phone ports?
>
> Cheers
>
> Vega
> -- next part --
> An HTML attachment was scrubbed...
> URL:
> 
>
>
Pls make sure you have configured the ephone-dns to be octo-line

thanks,
/Baktha

--
>
> Message: 2
> Date: Wed, 22 Feb 2012 12:35:57 +
> From: Rynard Coetzee 
> To: "ccie_voice@onlinestudylist.com" 
> Subject: [OSL | CCIE_Voice] cBarge in CME SRST
> Message-ID:
>
>  <97D19F256B859A48A6A93E8A9210E52A17AF8BBE@BYTESEXCH2K10N1.bytes.local>
> Content-Type: text/plain; charset="us-ascii"
>
> Hi All
> I have an issue to get the cBarge to work when my H323 GW goes into SRST
> ,the shared line shows up on both phones ,but when I have an active call on
> one phone ,I don`t see the number on the other phone ,and the other phone
> does not go into remote in use state when I press the shared line button. I
> have privacy turned off under the ephones and also under the telephony
> service. Also my CFB is registered to the router when in srst mode ,I am
> able to make a normal ad-hoc conference when in srst mode.
> Any ideas ?
> Regards
> Rynard
> -- next part --
> An HTML attachment was scrubbed...
> URL:
> 
>
> --
>
> Message: 3
> Date: Wed, 22 Feb 2012 13:58:15 +0100
> From: "Farkas P?ter" 
> To: Rynard Coetzee 
> Cc: "ccie_voice@onlinestudylist.com" 
> Subject: Re: [OSL | CCIE_Voice] cBarge in CME SRST
> Message-ID: <770ca7cc57a8.4f44f...@sch.bme.hu>
> Content-Type: text/plain; charset=us-ascii
>
> Have you tried to turn off privacy and enable remote-in-use sofktkey
> through an ephone-template attached to the ephone? Privacy setting on
> ephone has a bug in SRST mode.
>
> Peter
> - Original Message -
> From: Rynard Coetzee 
> Date: Wednesday, February 22, 2012 1:51 pm
> Subject: [OSL | CCIE_Voice] cBarge in CME SRST
> To: "ccie_voice@onlinestudylist.com" 
>
>
> > Hi All
> >  I have an issue to get the cBarge to work when my H323 GW goes into
> SRST ,the shared line
> > shows up on both phones ,but when I have an active call on one phone ,I
> don`t see the number on
> > the other phone ,and the other phone does not go into remote in use
> state when I press the
> > shared line button. I have privacy turned off under the ephones and also
> under the telephony
> > service. Also my CFB is registered to the router when in srst mode ,I am
> able to make a normal
> > ad-hoc conference when in srst mode.
> >  Any ideas ?
> >  Regards
> >  Rynard
> > ___
> >  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] CCIE_Voice Digest, Vol 72, Issue 115

[Baktha Muralidharan]

>
> Hi Randall


1. I think the solutions guide recommends that you run auto qos on an *
unused* interface, just to be safe.
2.  As you know, the 75% refers to the value of the [drop] threshold. Since
cos 5 is associated with 3rd threshold, as such, you would configure
threshold 3 to be 75%. However, the recommended approach is to move the cos
5 to threshold 1 and then configure threshold 1 to be 75%. The theory
behind this is somewhat complex (Vik explains this in detail in the
bootcamp). It goes something like this-

Threshold 3 is kind of the "percentage of buffer contributed by this queue
to the overall buffer space" and threshold 4 specifies the 3rd threshold
for the queue in question.

thanks,
/Baktha



> --
>
> Message: 2
> Date: Sun, 19 Feb 2012 11:15:30 -0800 (PST)
> From: Randall Crumm 
> To: Online Study 
> Subject: [OSL | CCIE_Voice] New lab #2 - 3750 Qos
> Message-ID:
><1329678930.97124.yahoomail...@web164603.mail.gq1.yahoo.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> HI,
> I have a question on Qos
> 1. The question wants COS 3 i n the 2nd egress queue, 3rd threshold and
> COS 5 in the 1st egress queue?3rd threshold.
> This is done after running auto qos on the phone switchport. No question
>
>
> 2. COS 5 ?traffic sent to SA-GW(queue set 2) should be dropped if queue is
> 75% full.?
> Can someone please explain the answer to me. The DSG is not clear to me.
>
> Thanks,
>
>
> ?
> Cheers,
> Randall
> -- next part --
> An HTML attachment was scrubbed...
> URL:
> 
>
> --
>
>

Hi Emanuel,

  - does the call drop even if you select an internal (on-net) extension?
  - assume you have configured the CSS on the RDP?

thanks,
/Baktha




> Message: 3
> Date: Sun, 19 Feb 2012 17:17:17 -0200
> From: Emanuel Damasceno 
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] MVA - Phone hangs up when trying to make a
>call.
> Message-ID:
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hey guys,
>
> I've configured everything (I suppose...) RDP, RD, MVA on SP... Everything
> seems to be in order. It welcomes me, asks to dial a password, yadda yadda,
> when I ask to make a call by pressing 1, I dial the numbers and when I
> press # the connection drops.
>
> Here is my config:
> HQ-RTR
> application
>  service mva http://10.10.210.10:8080/ccmivr/pages/IVRMainpage.vxml
> !
> dial-peer voice 5999 pots
>  service mva
>  incoming called-number 5999
> !
> dial-peer voice 5000 voip
>  destination-pattern 5...
>  voice-class codec 1
>  voice-class h323 1
>  session target ipv4:10.10.210.11
>  no vad
> dial-peer voice 5001 voip
>  preference 1
>  destination-pattern 5...
>  voice-class codec 1
>  voice-class h323 1
>  session target ipv4:10.10.210.10
>  no vad
>
> The remote Profile and Remote Destination Profile are correct, otherwise it
> wouldn't ask me to put the PIN straight up. It would ask me first for my
> number then PIN.
> Any ideas?
>
> *Emanuel Damasceno*
> CCNP Voice
> -- next part --
> An HTML attachment was scrubbed...
> URL:
> 
>
> --
>
> Message: 4
> Date: Sun, 19 Feb 2012 11:16:44 -0800 (PST)
> From: Randall Crumm 
> To: romain mullier 
> Cc: Online Study 
> Subject: Re: [OSL | CCIE_Voice] BACD issue new labs #2
> Message-ID:
><1329679004.50928.yahoomail...@web164601.mail.gq1.yahoo.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> lol
>
> You mean load up some more. I have to wake up at 5AM since I am on the
> west coast.
> ?
> Cheers,
> Randall
>
>
>
> 
>  From: romain mullier 
> To: Randall Crumm 
> Cc: Online Study 
> Sent: Sunday, February 19, 2012 11:09 AM
> Subject: Re: [OSL | CCIE_Voice] BACD issue new labs #2
>
>
> Randall,
>
> You defined the name of your sevice as "app-b-acd-aa" yet your dial-peer
> is invoking "service aa" which does not exist.
> Time to take a coffee break ;-)
>
> Romain
>
>
>
> On Sun, Feb 19, 2012 at 12:59 PM, Randall Crumm  wrote:
>
> Hi I am trying to call x4000 and have BACD forward to ephone-hunt list.
> >
> >
> >When i call 4001 it works. When I call 4000 I get a busy signal .
> >
> >
> >Any thoughts are appreciated.
> >
> >
> >application
> >?service app-b-acd-aa
> >? param voice-mail 4600
> >? paramspace english index 1
> >? param max-time-call-retry 700
> >? param service-name app-b-acd
> >? param number-of-hunt-grps 1
> >? param drop-through-option 1
> >? paramspace english language en
> >? param handoff-string app-b-acd-aa
> >? param max-time-vm-retry 2
> >? paramspace english location flash:
> >? param aa-pilot 4000
> >? param second-greeting-time 60
> >? param welcome-prompt _bacd_welcome.au
> >? param call-retry-timer 15
> >?!
> >?service app-b-acd
> >? param queue-len 15
> >? param aa-hunt1 4123
> >? param queue-manager-debugs 1
> >? param number-of-hunt-grps 1
> >
> >
> >dial-peer voice 222 voip
> >?service aa
> >?destinat

[OSL | CCIE_Voice] BACD issue new labs #2

[Baktha Muralidharan]

the service comman dunder the dial-peer is wrong-- should be app-b-acd-aa


dial-peer voice 222 voip
?service *aa*
?destination-pattern 4000
?session target ipv4:10.10.110.3
?incoming called-number 4000
?dtmf-relay h245-alphanumeric
?codec g711ulaw
?no vad

thanks,
/Baktha

 Message: 3
Date: Sun, 19 Feb 2012 09:59:26 -0800 (PST)
From: Randall Crumm 
To: Online Study 
Subject: [OSL | CCIE_Voice] BACD issue new labs #2
Message-ID:
   <1329674366.76349.yahoomail...@web164611.mail.gq1.yahoo.com>
Content-Type: text/plain; charset="iso-8859-1"

Hi I am trying to call x4000 and have BACD forward to ephone-hunt list.

When i call 4001 it works. When I call 4000 I get a busy signal .

Any thoughts are appreciated.

application
?service app-b-acd-aa
? param voice-mail 4600
? paramspace english index 1
? param max-time-call-retry 700
? param service-name app-b-acd
? param number-of-hunt-grps 1
? param drop-through-option 1
? paramspace english language en
? param handoff-string app-b-acd-aa
? param max-time-vm-retry 2
? paramspace english location flash:
? param aa-pilot 4000
? param second-greeting-time 60
? param welcome-prompt _bacd_welcome.au
? param call-retry-timer 15
?!
?service app-b-acd
? param queue-len 15
? param aa-hunt1 4123
? param queue-manager-debugs 1
? param number-of-hunt-grps 1

dial-peer voice 222 voip
?service aa
?destination-pattern 4000
?session target ipv4:10.10.110.3
?incoming called-number 4000
?dtmf-relay h245-alphanumeric
?codec g711ulaw
?no vad
!
dial-peer voice 1 pots
?incoming called-number .
?direct-inward-dial
!
!
!
telephony-service
?srst mode auto-provision all
?em logout 0:0 0:0 0:0
?max-ephones 8
?max-dn 8
?ip source-address 10.10.202.1 port 2000
?time-zone 21
?time-format 24
?max-conferences 8 gain -6
?moh music-on-hold.au
?transfer-system full-consult
?create cnf-files version-stamp 7960 Feb 19 2012 22:18:09
!
!
ephone-dn ?1
?number 4001
?label 4001
?description +442077964001
?name +442077964001
!
!
ephone ?1
?mac-address 0024.142E.76A9
?button ?1:1
!
!
ephone-hunt 1 longest-idle
?pilot 4123
?list 4001
!
!
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[OSL | CCIE_Voice] B-ACD max-time-call-retry timer

[Baktha Muralidharan]

Hello all,

per the B-ACD guide, the definition of max-time-call-retry is as follows-

*If the amount of time a call is in the queue exceeds the limit set by the
param max-time-call-retry command, the call is routed to the alternate
destination set with the param voice-mail command. If the param
max-time-vm-retry command is set to a number higher than one, the
call-queue service retries to connect that number of times.*

I have the following B-ACD config-

application
 service app-b-acd-aa
  param voice-mail 4220
  paramspace english index 1
  param *max-time-call-retry 30*
  param service-name app-b-acd
  param number-of-hunt-grps 1
  param drop-through-option 2
  paramspace english language en
  param handoff-string app-b-acd-aa
  param *max-time-vm-retry 0*
  paramspace english location flash:
  param aa-pilot 4019
  param second-greeting-time 10
  param welcome-prompt _bacd_welcome.au
  param call-retry-timer 5
 !
 service app-b-acd
  param queue-len 15
  param number-of-hunt-grps 1
  param aa-hunt2 4111
  param queue-manager-debugs 1

Yet, calls hit voicemail ONLY after a minute (60 secs).

Any hints ? Anybody able to get max-time-call-retry to work?

Thanks in advance,
/Baktha
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[OSL | CCIE_Voice] : E1 configuration (Eliot Ngwa)

[Baktha Muralidharan]

please check to make sure that network-clock synchronization is configured.

  "show network-clock"

thanks,
/Baktha



>
> Message: 7
> Date: Tue, 14 Feb 2012 05:22:33 -0800 (PST)
> From: muhammad nouman 
> To: "michael.se...@compucom.com" ,
>"ccie_voice@onlinestudylist.com" 
> Subject: Re: [OSL | CCIE_Voice] E1 configuration (Eliot Ngwa)
> Message-ID:
><1329225753.3682.yahoomail...@web120705.mail.ne1.yahoo.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
>
>
> Hi All,
> ?
> I have connected both E1 back to back with PRI crossover but I am getting
> following error on both side.
> ?
> Please let me know is this normal or it will create problem.
> ?
> HQ-Router#sh controllers e1
> E1 0/3/0 is up.
> ? Applique type is Channelized E1 - balanced
> ? No alarms detected.
> ? alarm-trigger is not set
> ? Version info Firmware: 20090113, FPGA: 20, spm_count = 0
> ? Framing is CRC4, Line Code is HDB3, Clock Source is Internal.
> ? Data in current interval (202 seconds elapsed):
>  0 Line Code Violations, 0 Path Code Violations
>  3 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
>  3 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail
> ? Total Data (last 2 15 minute intervals):
>  3 Line Code Violations, 5 Path Code Violations,
>  30 Slip Secs, 0 Fr Loss Secs, 3 Line Err Secs, 0 Degraded Mins,
>  32 Errored Secs, 1 Bursty Err Secs, 0 Severely Err Secs, 0 Unavai
> HQ-Router#
>
> PSTN-FRS#sh controllers e1
> E1 0/2/0 is up.
> ? Applique type is Channelized E1 - balanced
> ? No alarms detected.
> ? alarm-trigger is not set
> ? Version info Firmware: 20090113, FPGA: 20, spm_count = 0
> ? Framing is CRC4, Line Code is HDB3, Clock Source is Internal.
> ? Data in current interval (315 seconds elapsed):
>  0 Line Code Violations, 0 Path Code Violations
>  4 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
>  4 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs
> ? Total Data (last 2 15 minute intervals):
>  3 Line Code Violations, 3 Path Code Violations,
>  31 Slip Secs, 0 Fr Loss Secs, 2 Line Err Secs, 0 Degraded Mins,
>  33 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 2 Unavail
> Secs
> ?
> ?
> I am also not able to register this E1 with MGCP, please help me
> ?
> Thanks
> ?
> Nomi
>
> ?
>
> 
>  From: "michael.se...@compucom.com" 
> To: ccie_voice@onlinestudylist.com
> Sent: Saturday, 31 December 2011 5:40 AM
> Subject: [OSL | CCIE_Voice] E1 configuration (Eliot Ngwa)
>
> I read in the thread below "I am using simple crossover cable (Ethernet
> crossover)".? This cable will not work.? You need a T1 cross over cable:?
> http://www.google.com/search?q=t1+crossover+cable&hl=en&prmd=imvns&tbm=isch&tbo=u&source=univ&sa=X&ei=3gP-Tp2AIpS5twf-ufXPBg&sqi=2&ved=0CF4QsAQ&biw=1072&bih=804
>
> !!!
> If your using an Ethernet cross over it won't work need T1 cross over
> 1-->4, 2-->5, 4-->1, 5-->2
>
> http://www.ebay.com/itm/T1-Crossover-cable-3FT-/160570999135?pt=LH_DefaultDomain_0&hash=item2562c7015f
>
> Usually you can pick one up at local computer store or ebay real cheap
> depending on the length.
>
> Hope this helps.
> Michael Sears
>
>
>
> ___
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> visit www.ipexpert.com
>
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>
> End of CCIE_Voice Digest, Vol 72, Issue 80
> **
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[OSL | CCIE_Voice] Passed

[Baktha Muralidharan]

Congratulations KP!!!

Message: 2
Date: Mon, 13 Feb 2012 12:30:59 -0900
From: CCIEVoiceKP 
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Passed
Message-ID:
   
Content-Type: text/plain; charset="iso-8859-1"

Just got the results this morning ... passed!  CCIE Voice #34696!  After 5
attempts, what a huge huge huge relief!  Thanks to everyone on the forum
for your participation.  Even if we never corresponded directly, I've
learned a ton from others experiences!  Keep up the good work!

KP
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[OSL | CCIE_Voice] PSTN Router - hairpin call or from HQ to SC

[Baktha Muralidharan]

>
> Jawwad
>
can you provide some details on what you want to accomplish?

So, a call originates from an HQ phone, and is routed via PSTN to an SC
phone?
then, can't you set up call-forward all on the called SC phone to hair the
call back to the calling HQ?

thanks,
/Baktha



> Message: 2
> Date: Sat, 11 Feb 2012 14:44:44 -0800 (PST)
> From: Jawwad Salahuddin 
> To: ccie voice 
> Subject: [OSL | CCIE_Voice] PSTN Router - hairpin call or from HQ to
>SC
> Message-ID:
><1329000284.33852.yahoomail...@web161401.mail.bf1.yahoo.com>
> Content-Type: text/plain; charset="us-ascii"
>
> Hello all,
>
> Can any one advice how can I do a hairpin call from PSTN router.
>
> Scenario HQ call SC via PSTN. What is required on PSTN Router.
>
> Thank you and Regards,
> Jawwad
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>
>
> End of CCIE_Voice Digest, Vol 72, Issue 71
> **
>
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[OSL | CCIE_Voice] DHCP Timeout

[Baktha Muralidharan]

- not sure if "utils csa disable" will help since it seems DHCP server IS
doling out IP addresses
- in fact, prbably all is ok on UCM, if it is giving out IP addresses to
BR1 phones, even though they are on a different subnet
- you could turn on "debug ip dhcp server events" on HJQ router, to see if
broadcast messages requesting IP addr are going out to UCM
- make sure there are no vlan restrictions on the trunk between switch and
HQ router.
- make sure no DHCP snooping is going on on the switch
- make sure DHCP service is running on HQ router

thanks,
/Baktha

On Fri, Feb 10, 2012 at 11:16 AM, wrote:

> Send CCIE_Voice mailing list submissions to
>ccie_voice@onlinestudylist.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
>http://onlinestudylist.com/mailman/listinfo/ccie_voice
> or, via email, send a message with subject or body 'help' to
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>
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>ccie_voice-ow...@onlinestudylist.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of CCIE_Voice digest..."
>
>
> Today's Topics:
>
>   1. Re: DHCP Timeout (Ramy Abdelrahim)
>   2. Re: DHCP Timeout (Rrcrumm)
>   3. Re: DHCP Timeout (Eliot Ngwa)
>   4. Re: DHCP Timeout (Kevin Spicer)
>
>
> --
>
> Message: 1
> Date: Fri, 10 Feb 2012 15:18:28 +
> From: Ramy Abdelrahim 
> To: 
> Cc: ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] DHCP Timeout
> Message-ID: 
> Content-Type: text/plain; charset="windows-1256"
>
>
> CUCM is pingable from  both the switch and the HQ-RTR. HQ-RTR#ping
> 10.10.210.10 source 10.10.200.3
> Type escape sequence to abort.Sending 5, 100-byte ICMP Echos to
> 10.10.210.10, timeout is 2 seconds:Packet sent with a source address of
> 10.10.200.3!Success rate is 100 percent (5/5), round-trip min/avg/max =
> 1/2/4 ms
>
> Date: Fri, 10 Feb 2012 08:17:42 -0600
> Subject: Re: [OSL | CCIE_Voice] DHCP Timeout
> From: whl...@gmail.com
> To: ramyoth...@hotmail.com
> CC: ccie_voice@onlinestudylist.com
>
> OK so BR1 is working but HQ is not.
>
> Difference is that BR1 the switch ports have direct access to the routing,
> they are clearly being routed.
>
> My guess is that the voice vlan on HQ is not able to reach the CUCM.
>
> See if you can ping the CUCM from the switch.  If this does not work, then
> you will have to find your routing issue.
>
>
> 2012/2/10 Ramy Abdelrahim 
>
>
>
>
>
> Dear All,
> I faced a scenario on workbook 2 that requests to have HQ and BR1 phones
> acquire their IP addresses from UCM-PUB. What happened was BR1 phones were
> able to get IP addresses from the UCM-PUB but HQ phones were not. The
> Switch and HQ router configuration is as follows. I appreciate if anyone
> can help on that.
>
> NOTE: The UCM-PUB is pingable from the switch and the HQ-RTR.
> Switch:
> vlan 10 name DATA!vlan 20
>  name PHONES!vlan 30 name SERVERS!interface fastethernet 1/0/1 -- To HQ
> router switchport trunk encapsulation dot1q switchport mode trunk
>  switchport trunk native vlan 10!interface fastethernet 1/0/2 -- HQ Phone
> 1 switchport access vlan 10 switchport mode access switchport voice vlan 20
> spantree portfast
>
> !///
> HQ-RTR:
> interface fastethernet 0/0.10
>  encapsulation dot1q 10 native ip address 10.10.100.1
> 255.255.255.0!interface fastethernet 0/0.20 encapsulation dot1q 20 ip
> address 10.10.200.3 255.255.255.0
>  ip helper-address 10.10.210.10!interface fastethernet 0/0.30
> encapsulation dot1q 30 ip address 10.10.210.1 255.255.255.0
>
> !///
>
> Thanks
>
> Ramy
>
> ___
>
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Re: [OSL | CCIE_Voice] No Audio on TEHO calls from H.323 to MGCP Gateways

[Baktha Muralidharan]

of course, another quick check is to hit the "i" button (or "?" button on
the phones) and see if the Rx/Tx counts are incrementing.

thanks,
/Baktha

On Thu, Feb 9, 2012 at 12:41 PM, Baktha Muralidharan
wrote:

> could you attatch outputs of
>
>   show voip rtp connections
>   show sccp connn
>   show call active voice brief
>
> thanks,
> /Baktha
>
>
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Re: [OSL | CCIE_Voice] No Audio on TEHO calls from H.323 to MGCP Gateways

[Baktha Muralidharan]

could you attatch outputs of

  show voip rtp connections
  show sccp connn
  show call active voice brief

thanks,
/Baktha

On Thu, Feb 9, 2012 at 12:00 PM, wrote:

> Send CCIE_Voice mailing list submissions to
>ccie_voice@onlinestudylist.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
>http://onlinestudylist.com/mailman/listinfo/ccie_voice
> or, via email, send a message with subject or body 'help' to
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>
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>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of CCIE_Voice digest..."
>
>
> Today's Topics:
>
>   1. Re: No Audio on TEHO calls from H.323 to MGCP Gateways (Vik Malhi)
>   2. Re: MVA problem (datucha123 datucha123)
>
>
> --
>
> Message: 1
> Date: Thu, 9 Feb 2012 08:34:10 -0800
> From: Vik Malhi 
> To: CCIEVoiceKP 
> Cc: ccie_voice@onlinestudylist.com, Ken Wyan 
> Subject: Re: [OSL | CCIE_Voice] No Audio on TEHO calls from H.323 to
>MGCPGateways
> Message-ID: <3c6e73ab-793c-46e1-93cc-b15a41d5a...@ipexpert.com>
> Content-Type: text/plain; charset="windows-1252"
>
> Are you using MTP and/or transcoders on the gateways?
>
>
> Vik Malhi ? CCIE #13890
> Managing Partner - IPexpert, Inc.
>
> Telephone: +1.810.326.1444 ext 420
> Fax: +1.810.454.0130
> Mailto: vma...@ipexpert.com
>
>
>
>
> On Feb 8, 2012, at 10:38 PM, CCIEVoiceKP wrote:
>
> > Yep, reloaded the GW, still no luck.  It's definitely a codec issue 
> I changed the Region seting to let HQ and BR1 communicate with g711u and
> viola ... works like a charm.  I set it back to g729r8 between hq and br1
> and no audio.
> >
> > My voip dial-peers to cucm bith contain voice-class codec 1 ... and that
> voice class has both g711u and 729r8 
> >
> > I'm running out of ideas .
> >
> > KP
> >
> > On Wed, Feb 8, 2012 at 9:30 PM, Ken Wyan  wrote:
> > Did you reload gateway?
> >
> > On Thu, Feb 9, 2012 at 9:18 AM, CCIEVoiceKP 
> wrote:
> > I'm making TEHO calls from BR1 Router (H323) to HQ (MGCP).  The calls
> connects and stays connected however there is no audio between the two
> endpoints.  If I make the call, again TEHO, in the other direction form HQ
> to BR1 the call connects and there is audio.  If I shut the HQ voice-port
> down and force the call out of the Br1 GW the call connects and there is
> audio.
> >
> > I have transcoders registered to CUCM on both gateways, they are in hte
> proper MRGs, MRGLs, and Device Pools.
> >
> > Has anyone ever run into this?  I sit the lab on riday and it seems
> there is always something that pops up :(
> >
> > KP
> >
> > ___
> > 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] MGCP troubleshooting

[Baktha Muralidharan]

Hello,

Unless  the number is not coming in "en bloc". In overlap-receiving mode,
digits would come in digit by digit.
can you post the output of "debug isdn q931"?

Thanks,
/Baktha


> Message: 4
> Date: Wed, 8 Feb 2012 17:03:32 +0400
> From: datucha123 datucha123 
> To: Jeferson Guardia 
> Cc: OSL Voice 
> Subject: Re: [OSL | CCIE_Voice] MGCP troubleshooting
> Message-ID:
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
> debug mgcp packets
>
> Trace in CUCM
>
> On Wed, Feb 8, 2012 at 3:35 PM, Jeferson Guardia  >wrote:
>
> > Hi,
> >
> > What are the techniques most used to perform MGCP troubleshooting?
> >
> > Yesterday I was doing a lab and had a router with pstn integration, it
> was
> > set for a CSS where my phones had visibility and significant digits = 4.
> > But whenever I would call out from the PSTN, I would get a second dial
> > tone, I would see the call kicking in thru debug isdn q931 but my phone
> > would simply not ring. Any ideas how to verify that possible behavior ?
> > Articles? tech guides?
> >
> > Thanks,
> >
> > --
> > Jeferson Guardia
> > CCIE #28157
> >
> > ___
> > 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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>
> End of CCIE_Voice Digest, Vol 72, Issue 44
> **
>
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[OSL | CCIE_Voice] OSL | CCIE_Voice] QOS LFI and BACD files

[Baktha Muralidharan]

Of course, if the question made you do class-based shaping, then, you don't
need to worry about the other links.
Only FRTS will cause the links to drop to default (i.e. 56kbps).
Always a good idea to do "show traffic" command, after configuring one
link, to see if the other link has defaulted to 56kbps speed.

/Baktha


On Tue, Feb 7, 2012 at 10:58 AM, wrote:

> Send CCIE_Voice mailing list submissions to
>ccie_voice@onlinestudylist.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
>http://onlinestudylist.com/mailman/listinfo/ccie_voice
> or, via email, send a message with subject or body 'help' to
>ccie_voice-requ...@onlinestudylist.com
>
> You can reach the person managing the list at
>ccie_voice-ow...@onlinestudylist.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of CCIE_Voice digest..."
>
>
> Today's Topics:
>
>   1. Re: QOS LFI and BACD files (Robert Schuknecht)
>   2. QOS LFI and BACD (Larry Stern)
>   3. Re: QOS LFI and BACD files (CCIEVoiceKP)
>
>
> --
>
> Message: 1
> Date: Tue, 7 Feb 2012 16:48:42 +0100
> From: "Robert Schuknecht" 
> To: "'AJ BG'" ,
>
> Subject: Re: [OSL | CCIE_Voice] QOS LFI and BACD files
> Message-ID: <00ed01cce5af$fad8a000$f089e000$@gmx.de>
> Content-Type: text/plain; charset="us-ascii"
>
> Hello AJ,
>
>
>
> regarding BACD, i would you the Bulid-In Scripts inside IOS.
>
>
>
> /Robert
>
>
>
> Von: ccie_voice-boun...@onlinestudylist.com
> [mailto:ccie_voice-boun...@onlinestudylist.com] Im Auftrag von AJ BG
> Gesendet: Tuesday, February 07, 2012 7:03 AM
> An: ccie_voice@onlinestudylist.com
> Betreff: [OSL | CCIE_Voice] QOS LFI and BACD files
>
>
>
> Hello,
>
>   1.   QOS question
>
> According to Vic, if you configure LFI for a subinterface in a hub and
> spoke
> environment, Your second sub interface will dopes its CIR to 56k. To solve
> this issue you should configure map-class for the second interface as well.
> I have tested this and confirmed the problem and the solution.
>
>  But if the interface bandwidth is not given to you, then in what rate do
> you configure the second map-class? What should be your CIR and MinCIR
> bandwidth?
>
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[OSL | CCIE_Voice] Timezone for UCM

[Baktha Muralidharan]

Hello

I used to wonder about it too :-) In the absence of specific requirement
for timezone on UCM as such, I believe that, if the requirement is in terms
of the time as it shows up on the phones, then, the folowing should suffice-

  1. configuring NTP on UCM
  2. configuring date/time group and
  3. setting the date/time group on the device pools to which the phones
belong

Thanks,
/Baktha

> Message: 2
> Date: Mon, 6 Feb 2012 20:20:36 +0400
> From: datucha123 datucha123 
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] Timezone for UCM
> Message-ID:
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hello,
>
> If we are told to synchronize CUCM Pub server with some NTP, do we need to
> set the correct Timezone for CUCM OS as well? Or just Date/Time Groups for
> Phones are enough?
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>
> End of CCIE_Voice Digest, Vol 72, Issue 31
> **
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[OSL | CCIE_Voice] Locations CAC

[Baktha Muralidharan]

Just to make sure, did you turn on AAR in service parameters?

Thanks,
/Baktha

Message: 4
> Date: Sun, 5 Feb 2012 17:03:46 -0200
> From: Emanuel Damasceno 
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] Locations CAC
> Message-ID:
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hello Experts,
>
> I am trying to set up an AAR scenario for my studies. I configured 2
> Locations, with unlimited bandwidth, but mandatory RSVP from HQ to BR2. I
> wanna use 5 concurrent calls, and I am also using g729 between sites. I
> added the MTP-HQ, and MTP-BR2 to CUCM, put them in a MRG, followed by MRGL,
> and referenced it in its respective device pool. Reset all the phones.
>
> So here is my config:
> HQ
> dspfarm profile 2 mtp
>  codec g729r8
>  rsvp
>  maximum sessions software 5
>  associate application SCCP
>
> interface Serial0/0/1:0.2 point-to-point
>  description TO BR2
>  bandwidth 768
>  ip address 10.10.112.1 255.255.255.0
>  ip ospf mtu-ignore
>  snmp trap link-status
>  frame-relay interface-dlci 202
>  class AutoQoS-FR-Se0/0/1:0-202
>  auto qos voip trust
>  frame-relay ip rtp header-compression
>  ip rsvp bandwidth 136
>
> BR2
> dspfarm profile 1 mtp
>  codec g729r8
>  rsvp
>  maximum sessions software 5
>  associate application SCCP
>
> interface Serial0/1/0:0.1 point-to-point
>  description to HQ
>  bandwidth 768
>  ip address 10.10.112.2 255.255.255.0
>  ip ospf mtu-ignore
>  snmp trap link-status
>  frame-relay interface-dlci 102 CISCO
>  class AutoQoS-FR-Se0/1/0:0-102
>  auto qos voip
>  frame-relay ip rtp header-compression
>  ip rsvp bandwidth 136
>
> The main problem is that on the FIRST call it already says "Not Enough
> Bandwidth", wasn't that supposed to happen if the 6th caller tried to make
> a call? I already set to TRUE in Service Parameters for Automated Alternate
> Routing, but it's not showing the "Not Enough Bandwidh, Rerouting" message.
> I haven't configured my Partitions and CSSs yet, but what's up with the
> first call not going through? Am I missing something?
>
> *Emanuel Damasceno*
> CCNP Voice
> -- next part --
> An HTML attachment was scrubbed...
> URL: 
>
> --
>
> ___
> CCIE_Voice mailing list
> CCIE_Voice@onlinestudylist.com
> http://onlinestudylist.com/mailman/listinfo/ccie_voice
>
>
> End of CCIE_Voice Digest, Vol 72, Issue 27
> **
>
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[OSL | CCIE_Voice] BACD Not Invoking service

[Baktha Muralidharan]

Hi William

- did you mean to configure a drop-through B-ACD? then you need
   "param drop-through-option 1"
- You probably want "param aa-hunt1 " instead of "param aa-hunt2 "
- make sure moh is enabled and working
- if it still doesn't work, "please turn ON debug voice application vxml"
and post the debug capture.

thanks,
/Baktha


From: William Affeldt 
To: Vik Malhi , Ashwani 
Cc: "ccie_voice@onlinestudylist.com" 
Subject: [OSL | CCIE_Voice]   BACD Not Invoking service
Message-ID:
   <1328402616.84156.yahoomail...@web114616.mail.gq1.yahoo.com>
Content-Type: text/plain; charset="utf-8"

I configured BACD today and could not get it working. The dial-peer never
actually invokes the application. It does hit the correct dial-peer in and
out though. Below is my exact configuration. Any help would be greatly
appreciated.
?
?
?
application
service app-b-acd-aa
param voice-mail 5003
paramspace english index 1
param max-time-call-retry 700
param service-name app-b-acd
param number-of-hunt-grps 1
paramspace english language en
param handoff-string app-b-acd-aa
param dial-by-extension-option 1
param max-time-vm-retry 2
paramspace english location flash:bacdprompts/
param aa-pilot 4110
param second-greeting-time 60
param welcome-prompt _bacd_welcome.au
param queue-manager-debugs 1000
param call-retry-timer 15
?!
service app-b-acd
param queue-len 15
param number-of-hunt-grps 1
param aa-hunt2 
param queue-manager-debugs 1
?!
!
!
dial-peer voice 222 voip
?service app-b-acd-aa
?destination-pattern 4110
?session target ipv4:10.10.110.3
?incoming called-number 4110
?dtmf-relay h245-alphanumeric
?codec g711ulaw
?no vad
!
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Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 72, Issue 16

[Baktha Muralidharan]

Message: 6
Date: Sat, 4 Feb 2012 15:16:56 -0800
From: "John McGaughey (jomcgaug)" 
To: 
Subject: [OSL | CCIE_Voice] CFUR does not work
Message-ID:
   <28cea477be98ed48af08a0b637a74bf40fe3e...@xmb-sjc-224.amer.cisco.com>
Content-Type: text/plain; charset="us-ascii"

Hi Vik/All



I'm working on Lab #4 of the new 5 labs.  Quesiton 9.2.  They are asking
you to configure CFUR on SiteB phone 2.  However this will not work
because of the RDP assigned to the phone.


Uncheck the "Mobile connect" checkbox in the RDP page and the CFUR call
should start working.
there is no requirement for  SNR call in the question, and so, only need to
support the mapping (to x3002) for the incoming call.

thanks,
/Baktha
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[OSL | CCIE_Voice] CUE Live Reply

[Baktha Muralidharan]

Was the voicemail left from a PSTN phone/number? If yes, make sure you have
a route for the number.
Keep in mind this number will NOT have the "9" prefix. So, you need a
dial-peer with destination pattern WITHOUT the 9.
Finally, remember CME uses "first match" algorithm to find the matching
dial-peer and NOT "best match"; so be careful in designing your dial-peers
and voice translation rules.

thanks,
/Baktha
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Re: [OSL | CCIE_Voice] Vol2 Lab8 Question5.6

[Baktha Muralidharan]

can't you use a class-map with "match-any" as follows?

  class-map match-any foo
 match access-group rtpports
 match ip dscp cs1

thanks,
/Baktha


> Message: 3
> Date: Sat, 4 Feb 2012 00:18:28 -0800 (PST)
> From: Vega Wong 
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] Vol2 Lab8 Question5.6
> Message-ID:
><1328343508.576.yahoomailclas...@web65902.mail.ac4.yahoo.com>
> Content-Type: text/plain; charset="utf-8"
>
> Hi experts
>
> I am working on question 5.6 in Vol 2 Lab 8. But I dont understand how the
> DSG explains the answers.
>
> Questions:
> HQ and BR1 - 768kbps
> PQ for 4 calls
> 5% for signalling
> mark the excess RTP traffic (failed RSVP) to 33%. (this type of RTP been
> marked as CS1 in earlier question)
> Ensure FRF.12 LFI is being used.
>
> The DSG is saying we should not use "auto qos voip trust" for branch 1,
> due to the limitation on HWIC-ESW. However, if I do that, the class-maps
> created will be matching the traffic based on the port.
> Here is what I am confused, one of the class-map created is matching the
> RTP traffic by the ports. Then if I need to "catch" the RTP that was marked
> as CS1 by CUCM, I will need to create another class-map right? but how does
> the policy-map know which class it should put the RTP traffic in? In
> another words, some of the RTP traffic will be matched by two different
> classes. Which class would take precedences?
>
> On the other hand, if we are not to trust the phone connecting to the
> HWIC, should we be creating a service policy and apply it as a
> service-policy input on the HWIC port to mark all the traffic coming in to
> the branch router?
>
> Can anyone shed some light on this one?
>
> Cheers
>
> Vega
>
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[OSL | CCIE_Voice] MVA question

[Baktha Muralidharan]

Did you try adding "direct-inward-dial" to your POTS dial-peer?

Not sure if it could service parameters, for if they are not set/enabled,
you won't even hear the MVW IVR prompts.

/Baktha
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[OSL | CCIE_Voice] Quick convention question

[Baktha Muralidharan]

Hello folks,

If we have the situation wherein COR requires that HQ phone not be
able to make international calls, and if  COR is to be observed,
during SRST as well, would it be ok to allow inter-site calls (say,
from a HQ phone to a BR2 phone)?  As you know, it could be an
international call to BR2 phone.

thanks in advance,
/Baktha
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[OSL | CCIE_Voice] Voiceview getting Authentication error in SRST

[Baktha Muralidharan]

Hello,

Apologies if this might have been discussed before.

As part of Mock Lab 1, I have CUE integrated with UCM. I also have
voiceview configured.  The authentication URL (enterprise parameter)
on call manager is configured as,

    http:///CCMCIP/authenticate.asp

In regular (non SRST) mode,

   - voicemail works (including MWI)
   - voiceview works (i.e. I can access inbox and also listen to messages)

However, when the phone that has mailbox on the CUE Module goes into SRST,

   - Voicemail, Messages button and MWI all work.
   - I can press the Services button on the phone and see the Inbox
   - However, when I select the message, I get Authenticatinon error.

Not sure if this is because the phone is trying to authenticate with
UCM, which is not accessible.
I am using call-manager-fallback; not sure how to configure alternate
authentication URL under there.

Any help (or pointers, if there is an existing thread) on this will be
much appreciated.

regards,
/Baktha
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Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 68, Issue 14

[Baktha Muralidharan]

Hello folks

I understand that the VPN issue that proctorlabs had been having is now
resolved.

I am now able to ping the PSTN WAN router from my 3750 switch, but not able
to ping any of the servers.
And, from the hardware VPN router, I am not able to ping even the PSTN WAN
router. I tried clearing the connection and re-establishing it, to no avail.

Anyone else having issues with connectivty ?

Any hints ?

Thanks in advance,
/Baktha
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Re: [OSL | CCIE_Voice] IP Expert Vouchers

[Baktha Muralidharan]

My apologies. I understand that this  forum is not appropriate for trading
(vRack sessions).

Regards,
/Baktha
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[OSL | CCIE_Voice] Looking to buy vRack sessions

[Baktha Muralidharan]

Hello,

I am looking to buy vRack sessions. please let me know if you have coupons
you would like to sell.

Thanks,
/Baktha
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[OSL | CCIE_Voice] Mobile Connect Vol 2 Lab 5

[Baktha Muralidharan]

Hello folks

I am having trouble getting Mobile connect to work and would appreciate any
tips on the same.

- I have RDP and RD configured for one of the phones (HQ Phone 2, at x4002,
specifically)
- I have owner ID set, line associated, rerouting CSS configured
- when I call from x3002 to x4002, after the delay (2000ms), the "mobile"
phone corresponding to x4002 does not  ring
- however, if I answer the call on x4002, I am then able to send the call to
mobile, by pressing the Mobility Softkey.

Any hints what I might be missing?

Thanks,
/Baktha
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Re: [OSL | CCIE_Voice] no "auto qos" command under Serial Interface

[Baktha Muralidharan]

Please disregard.. had to go under DLCI submode.

thanks!
/Baktha

On Sun, Feb 13, 2011 at 7:24 PM, Baktha Muralidharan
wrote:

> Hello,
>
> As part of Lab 10A, I tried to configure "auto qos voip fr-atm" under
> Serial 0/0/1:0.2. However, IOS CLI doesn't seem to like the command.
> Any hints on what is going on?
>
> I tried it on HQ,BR1 and BR2 routers. same result.
>
> thanks in advance,
> /Baktha
>
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[OSL | CCIE_Voice] no "auto qos" command under Serial Interface

[Baktha Muralidharan]

Hello,

As part of Lab 10A, I tried to configure "auto qos voip fr-atm" under Serial
0/0/1:0.2. However, IOS CLI doesn't seem to like the command.
Any hints on what is going on?

I tried it on HQ,BR1 and BR2 routers. same result.

thanks in advance,
/Baktha
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Re: [OSL | CCIE_Voice] Phone service doesn't show

[Baktha Muralidharan]

That was it!
yes, had checked Enterprise subscription!.
deleted that phone service and added another one WITHOUT the Enterprise
subscription.. now, I see the EM in the pull down menu.

thanks much,
/Baktha

On Wed, Feb 9, 2011 at 8:51 PM, Roger Carpio  wrote:

> Did you check the option "Enterprise Subscription" when you created the
> service?
>
> Regards,
> Roger Carpio.
>
>   On Wed, Feb 9, 2011 at 6:50 PM, Baktha Muralidharan <
> muralic...@gmail.com> wrote:
>
>>   Hello,
>>
>>
>>
>> As part of lab 9A, I configured phone service for extension mobility, with
>> service URL etc. I then restarted the EM service under “CM Servicability”
>> page.
>>
>> However, when I go into the Subscribe/Unsubscribe services page for a
>> phone or device profile,  I don’t see the [EM] service in the pull-down
>> menu.
>>
>> All I see for “Select a service” is “Intercomm calls”
>>
>>
>>
>> Any hints on what I am missing will be much appreciated.
>>
>>
>>
>> Thanks,
>>
>> /Baktha
>>
>> ___
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>> visit www.ipexpert.com
>>
>>
>
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Re: [OSL | CCIE_Voice] Phone service doesn't show

[Baktha Muralidharan]

yes, enable checked.

thanks.
On Wed, Feb 9, 2011 at 8:44 PM,  wrote:

> Did you check the enable check box?
> -Original Message-
> From: Baktha Muralidharan 
> Sender: ccie_voice-boun...@onlinestudylist.com
> Date: Wed, 9 Feb 2011 19:50:09
> To: 
> Subject: [OSL | CCIE_Voice] Phone service doesn't show
>
> ___
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> visit www.ipexpert.com
>
>
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[OSL | CCIE_Voice] Phone service doesn't show

[Baktha Muralidharan]

Hello,



As part of lab 9A, I configured phone service for extension mobility, with
service URL etc. I then restarted the EM service under “CM Servicability”
page.

However, when I go into the Subscribe/Unsubscribe services page for a phone
or device profile,  I don’t see the [EM] service in the pull-down menu.

All I see for “Select a service” is “Intercomm calls”



Any hints on what I am missing will be much appreciated.



Thanks,

/Baktha
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Re: [OSL | CCIE_Voice] SIP Phones on CME - TFTP: Looking for XMLDefault.cnf.xml.sgn (Kent Noorda - Q)

[Baktha Muralidharan]

the "upgrade" command goes under "voice register global"

/Baktha
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Re: [OSL | CCIE_Voice] Converting from SIP to SCCP, with CME (BR2), for Lab 3A, over ezVPN

[Baktha Muralidharan]

Folks,

thanks alot for your replies.

Good to know that there are ways to get the phone converted.

It will still be good to understand why it fails in the scenario involving
CME, ezVPN etc. Is it an issue with CME, or ezVPN or that I had different
"source" address under telephony-service and "voice register global" ?

thanks in advance,
/Baktha
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[OSL | CCIE_Voice] Converting from SIP to SCCP, with CME (BR2), for Lab 3A, over ezVPN

[Baktha Muralidharan]

Hi folks,



As part of Lab 3A (Volume 1),  I had to get  one phone at home registered as
SCCP and another, as SIP, with CME. I have two 7960s at home, both of which
were SIP phones earlier. I was able to get one of the 7960s registered as a
SIP phone. However, I could not get the 2nd 7960 to register as a SCCP
phone.



Shouldn’t the phone request SEP config file, after failing to
get SIP.cnf.xml ? Any hints?



Could the following have been my issue-



-  As you know, phones at home get their IP addresses from the ezVPN
router.

-  the source address for SCCP is configured  (as required by the
Lab) to be 10.10.110.3 (Loopback IP address on BR2)

-  the source address for SIP is configured to be  10.10.202.1 (VLAN
400) (also, as required by the Lab)

-  I had “Alternate TFTP Server” configured on the phones as *
10.10.110.3*



Do I want 2nd phone’s “Alternate TFTP Server”  set to VLAN400, given that is
the source address configured under “voice register global” ?

[My session expired before I got a chance to try it L]


Thanks a lot in advance,

/Baktha
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[OSL | CCIE_Voice] CIPC Won't register with CME

[Baktha Muralidharan]

Hi folks

just occured me to check--  can we configure CME with hard phones that are
located at home, across hardware (ez) VPN ?

thanks,
/Baktha
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[OSL | CCIE_Voice] CME GUI looks weird

[Baktha Muralidharan]

Hi folks

As part of Lab 3A, I configured for GUI access into the CME, as follows

  ip http server
  ip http authentication local
  ip http path flash:/GUI

  and under telephony-service

 web admin system name admin pass cisco


After this, I go into browser, and try to access the ccme.html page, I first
see the window go grey as expected, then see "View window" displayed at the
top left corner, and then the page comes up with the tabs "Configure",
"Voice Mail", "Administration", "Report" and "Help". However, the screen
looks weird and pressing the mouse on the tabs do not show the pull down
menus. I have pop-up blocker turned OFF on my browser. tried Mozilla as well
as IE, to no avail.

I then looked into flash:/GUI/CME_GUI_README.TXT, and on the first line, it
says "CME GUI Version 7.0.0.1".

any hints?

Thanks in advance,
/Baktha
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[OSL | CCIE_Voice] CIPC won't register

[Baktha Muralidharan]

Hello folks,

As part of lab 3A, I configured an (SCCP) ephone on BR2 (CME) router, with
mac address and type. On CIPC running on my PC, I set the tftp address to
pint to the loopback on the BR2 router. The CIPC won't register. I see the
display "Unprovisioned" on the CIPC screen.  On CME, turning "debug tftp
eve", I see that CIPC seems to be trying to open CTLSEP02004c4f4f50.tlv".
Seeing this, I tried to configure "device-security-mode none", but this
command doesn't seem to be available/accepted.

Any hints? Has anyone else run into this?

CME router is running CME 7.0.1 and IOS 12.4(22)T

thanks,
/Baktha
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