Re: [OSL | CCIE_Voice] ISDN Channel not available :D

[Steffen Bruening]

Maybe you configured more channels for the controller as available on the
PSTN Router Inbound calls are taking the first channel, outgoing the last
channel.

As far as I remember the HQ router on proctorlabs has only 6 channels, site
B and C have 4 channels.

Am Donnerstag, 10. Januar 2013 schrieb Abel ... :

> Try inbound from the 911 line and check the channel used.
>
>
> On Wed, Jan 9, 2013 at 6:52 PM, Nicolas MICHEL wrote:
>
> Inbound calls are working perfectly :(
>
>
>
> *Jan  9 08:59:43.227: ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8 callref =
> 0x0084
>Bearer Capability i = 0x8090A2
>Standard = CCITT
>Transfer Capability = Speech
>Transfer Mode = Circuit
>Transfer Rate = 64 kbit/s
>Channel ID i = 0xA98381
>Exclusive, Channel 1
>Progress Ind i = 0x8583 - Origination address is non-ISDN
>Display i = 'Emergency Services'
>Calling Party Number i = 0x0080, '911'
>Plan:Unknown, Type:Unknown
>Called Party Number i = 0xA1, '2123945002'
>Plan:ISDN, Type:National
> *Jan  9 08:59:43.251: ISDN Se0/0/0:23 Q931: TX -> CALL_PROC pd = 8
>  callref = 0x8084
>Channel ID i = 0xA98381
>Exclusive, Channel 1
> HQ-RTR(config-if)#
> *Jan  9 08:59:43.267: ISDN Se0/0/0:23 Q931: TX -> ALERTING pd = 8  callref
> = 0x8084
> *Jan  9 08:59:44.191: ISDN Se0/0/0:23 Q931: RX <- DISCONNECT pd = 8
>  callref = 0x0084
>Cause i = 0x8290 - Normal call clearing
> *Jan  9 08:59:44.195: ISDN Se0/0/0:23 Q931: TX -> RELEASE pd = 8 callref =
> 0x8084
> *Jan  9 08:59:44.207: ISDN Se0/0/0:23 Q931: RX <- RELEASE_COMP pd = 8
>  callref = 0x0084
> HQ-RTR(config-if)#
>
>
>
>
>
>
>
>
>
> Le Wednesday, January 09, 2013 11:50:14 PM, Abel ... a écrit :
>
> Try an inbound call from the pstn and run your debugs. Check the
> channel used in that call. That test is to discard L1/L2 issues.
>
> On Jan 9, 2013 6:21 PM, "Cory Gray"  > wrote:
>
> I will say this.  This happened to me once using MGCP and could
> never figure
> it out.  I just assumed "something was messed up" and could not
> find an
> answer in over 2 hours of troubleshooting.  If you ever figure it out,
> please let me know.
>
> -Original Message-
> From: ccie_voice-bounces@**onlinestudylist.com
> 
> [mailto:ccie_voice-bounces@**onlinestudylist.com
> ] On Behalf Of
> Nicolas MICHEL
> Sent: Wednesday, January 09, 2013 12:49 PM
> To: Bill
> Cc: OSL Voice
> Subject: Re: [OSL | CCIE_Voice] ISDN Channel not available :D
>
> Hi Guys,
>
>
> no the HQ is H323 otherwise I would not change the ISDN bchan
> order in the
> IOS but rather in the CUCM interface.
>
> the ISDN status shows Multi Frame Establish, all is all right. I
> can have
> incoming calls to the HQ but not outgoing calls to the Local and
> LD route
> pattern.
>  From what I remember, International calls are working great :(
>
>
> Next time I'll debug Q931 with detail enabled
>
>
> Thanks for the help :)
>
>
> Nic
>
>
>
>
>
>
>
> Le 1/9/2013 11:41 AM, Bill a écrit :
> > So is you hq a mgcp? If so it looks like you are missing your isdn
> > bind-l3 ccm command
> >
> > Is it h323 or maybe stand alone cme with just a plan pri?
> >
> > If h323 the does it show status in gateways? Does it show
> unknown unknown
> instead of unknown and ip?
> >
> > Try to give us more and clearer information because right now
> all we know
> is pri, but most likely hq is a gateway for CUCM and we don't know
> what
> type.  We don't know if this is your lab, rack rental or r
>
>
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Re: [OSL | CCIE_Voice] Connection Monitor Duration for SRST testing

[Steffen Bruening]

I always put it down to 10s, not less because sometimes the wan connection
is flapping/very low and so the phones flapping between cucm and srst.

Am Mittwoch, 9. Januar 2013 schrieb Pixar Perfect :

>  Hello,
>
> is it advisable to bring down the Connection Monitor Duration under Device
> Pool to a low value like 30 seconds to expedite the SRST testing in the
> lab? Does it affect grading if we happen to put it back to default?
>
> thx
>
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Re: [OSL | CCIE_Voice] Gatekeeper in Single zone : Call Processing engine prioritize

[Steffen Bruening]

You can define a gw-priority for the trunks to each endpoint.

Am Mittwoch, 9. Januar 2013 schrieb Ramcharan Arya :

> Hello,
>
> I have a small query about Gatekeeper in single zone configuration it is
> prefer PUB as call processing ( as per ARQ message).
>
> if lab requirement says make subscriber as call processing engine then it
> does not meet requirement.
>
> Does any one know the answer what is the solution of this tricky situation.
>
> Regards,
> Ramcharan Arya
> CCIE # 28926
>
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Re: [OSL | CCIE_Voice] Remote access on lab.

[Steffen Bruening]

Both is available on the desktop (in Brussels)

Am Mittwoch, 9. Januar 2013 schrieb Abdullin Kamil :

> Hi men,
> On lab what program is used for remote access on UCCX?
> windows remote desktop, VNC or any other?
> Thanks!
>
>
>
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Re: [OSL | CCIE_Voice] VOL2 lab 1 - HQ Phone cannot register ? WTH

[Steffen Bruening]

Hi Nicolas,

I had this problem too on some pods, where only one of the HQ phones want
to register, same for site B, dhcp addressing is working but no
registration occurs. Because I had no physical access on the phones to
check what the display is showing ( I guess something like registration
rejected), I tried several things to avoid this issue for the next session.

In most cases it has worked when I used completly different dhcp ranges in
the subnet, e.g 200.70 - 200.80.

When 7960s had problems to register it helps to configure them manually in
the cucm. Afterwards they are shown as registered.

In some circumstances it only had solved the issue when I configured a dhcp
pool on the router (for siteB probs) or on the HQ Sw for the HQ phones.





Am Dienstag, 8. Januar 2013 schrieb Nicolas MICHEL :

> Hey guys.
>
> I m in trouble with the phone connected to Fa1/0/23 of the 3750 in HQ.
>
> Vlan assignement are OK:
> HQ-3750#sh run int fa1/0/23
> Building configuration...
>
> Current configuration : 113 bytes
> !
> interface FastEthernet1/0/23
>  switchport access vlan 10
>  switchport voice vlan 20
>  spanning-tree portfast
> end
>
> HQ-3750#sh run int fa1/0/1
> Building configuration...
>
> Current configuration : 153 bytes
> !
> interface FastEthernet1/0/1
>  switchport trunk encapsulation dot1q
>  switchport trunk native vlan 10
>  switchport mode trunk
>  speed 100
>  duplex full
> end
>
> HQ-3750#sh int trunk
>
> PortMode Encapsulation  StatusNative vlan
> Fa1/0/1 on   802.1q trunking  10
>
> PortVlans allowed on trunk
> Fa1/0/1 1-4094
>
> PortVlans allowed and active in management domain
> Fa1/0/1 1,10,20,30
>
> PortVlans in spanning tree forwarding state and not pruned
> Fa1/0/1 1,10,20,30
>
>
>
>
> IP HELPER on the L3 Device is OK
>
> interface FastEthernet0/0.20
>  encapsulation dot1Q 20
>  ip address 10.10.200.3 255.255.255.0
>  ip helper-address 10.10.210.10
> end
>
>
> IP Phone can get an IP from the PUB (10.10.210.10)
>
> HQ-3750#sh cdp neigh fa1/0/23 det
> -
> Device ID: SEP0021A086825D
> Entry address(es):
>   IP address: 10.10.200.50
> Platform: Cisco IP Phone 7962,  Capabilities: Host Phone
> Interface: FastEthernet1/0/23,  Port ID (outgoing port): Port 1
> Holdtime : 178 sec
>
> Version :
> SCCP42.8-4-1S
>
> advertisement version: 2
> Duplex: full
> Power drawn: 6.300 Watts
> Power request id: 33373, Power management id: 1
> Power request levels are:6300 0 0 0 0
> Management address(es):
>
>
>
> the DHCP subnet is identical (except the IP range of course :) ) to the
> others DHCP pool which other phone use and can register to CUCM...
>
> I have no idea what is the problem here since it is a very basic problem
> 
>
> if anyone has experienced the same on POD 11 of PL ?
>
>
> Many thanks for your help on this
>
>
>
> __**_
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> 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] Hostname on the routers

[Steffen Bruening]

No don't do that, also don't add or remove domain names.

Am Sonntag, 6. Januar 2013 schrieb sanity insanity :

> hi Guys,
>
> I am just wondering if we are allowed to change host names on the routers
> . For example if the  changing the router name on R1 to HQ  and so too for
> the branches.
>
>
> -MJ
>
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Re: [OSL | CCIE_Voice] CUE CLI configuration

[Steffen Bruening]

You don't need to write a complete cue config, most is done with factory
defaults, only focus on integration, so ccn subsystem jtapi ( for cucm) ccn
subsystem sip (for cme and srst) ccn trigger for sip and jtapi and how to
create users and voicemail boxes.

Am Samstag, 5. Januar 2013 schrieb CCIEing :

> Hello Again guys,
>
> Return to CUE configuration using CLI  .
> So practicing CLI Imposes on us to know all the parameters for all
> applications, but there are a lot of parameters "like
> the autoattendant app", right?
>
> is there any way to remember them from inside the cli cue help, does any
> one have a difficulty to learn them .
>
> Appreciate your cooperation
>
> On Fri, Jan 4, 2013 at 2:12 PM, Chrysostomos Christofi <
> ch.christ...@logicom.net> wrote:
>
>  Sorry
>
> ** **
>
> Apologize wrong type
>
> ** **
>
> Correction:
>
> I heard from a lot guys that they had  troubles to access the cue through
> gui J
>
> ** **
>
> ** **
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Chrysostomos
> Christofi
> *Sent:* Παρασκευή, 4 Ιανουαρίου 2013 1:09 μμ
> *To:* Ahmad Taamneh; William Bell
> *Cc:* ccie_voice@onlinestudylist.com
> *Subject:* [OSL | CCIE_Voice] CUE CLI configuration
>
> ** **
>
> I heard from a lot guys that they had  troubles to access the cue through
> cli
>
> So we have to learn both ways
>
> ** **
>
> Also its more fast with cli either with cme or cucm integration
>
> ** **
>
> Regards
>
> Chrysostomos
>
> ** **
>
> ** **
>
> ** **
>
> *From:*
>
>
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Re: [OSL | CCIE_Voice] Any good blogs link on DB Replication Troubleshooting ?

[Steffen Bruening]

The dbreplication issues which you have at ipexpert/proctorlabs session you
can fix with an easy "utils dbreplication repair all", nothing more.

Am Samstag, 5. Januar 2013 schrieb Chrysostomos Christofi :

>  Hi Ram
>
> ** **
>
> ** **
>
> First check in PUB
>
> ** **
>
> utils network connectivity  To verify that PUB and SUB have connectivity**
> **
>
> ** **
>
> ** **
>
> 1. Sub: utils dbreplication stop (wait for it to complete before going to
> step 2.)
>
> 2. Pub: utils dbreplication stop (wait for it to complete before going to
> step 3.)
>
> 3. Pub: utils dbreplication reset all (monitor the dbstatus in RTMT for
> all 2’s.)
>
> 4. Reboot the cluster and re-verify in RTMT.
>
> ** **
>
> Final: 
>
> ** **
>
> utils dbreplication runtimestate
>
> ** **
>
> *From:* ccie_voice-boun...@onlinestudylist.com  'ccie_voice-boun...@onlinestudylist.com');> [mailto:
> ccie_voice-boun...@onlinestudylist.com  'ccie_voice-boun...@onlinestudylist.com');>] *On Behalf Of *Ramcharan Arya
> *Sent:* Σάββατο, 5 Ιανουαρίου 2013 8:19 μμ
> *To:* ccie_voice@onlinestudylist.com  'ccie_voice@onlinestudylist.com');>
> *Subject:* [OSL | CCIE_Voice] Any good blogs link on DB Replication
> Troubleshooting ?
>
> ** **
>
> Hello,
>
> ** **
>
> I am running on db replication issues very often.
>
> ** **
>
> PUB & Sub status are in these combinations 2:0, 3:0, 3:3, 4:4, 
>
> ** **
>
> Is any these structured procedure to fix these different state of db
> replications.
>
> ** **
>
> I have search IPX blogs but could not find information I was looking for.*
> ***
>
> ** **
>
> Can you someone please share detail procedures about how to fix DB
> replication between in following state:
>
> ** **
>
> Thanks & Regards,
> Ramcharan Arya
>
> ** **
>
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Re: [OSL | CCIE_Voice] Question for test class

[Steffen Bruening]

Its not allowed to bring any electronical "device" into the exam room.

Am Samstag, 5. Januar 2013 schrieb Marko Milivojevic :

> I would guess not. If you're concerned about it, you could open a case
> with the certifications support.
>
> --
> Marko Milivojevic - CCIE #18427 (SP R&S)
> Senior CCIE Instructor - IPexpert
>
> On Sat, Jan 5, 2013 at 9:31 AM, Chrysostomos Christofi
> > wrote:
> > Hi Marko
> >
> > I mean into the testing class for the ccie lab
> >
> >
> > -Original Message-
> > From: Marko Milivojevic [mailto:mar...@ipexpert.com ]
> > Sent: Σάββατο, 5 Ιανουαρίου 2013 7:27 μμ
> > To: Chrysostomos Christofi
> > Cc: Online Study (ccie_voice@onlinestudylist.com )
> > Subject: Re: [OSL | CCIE_Voice] Question for test class
> >
> > Into our class or the testing room?
> >
> > On Sat, Jan 5, 2013 at 5:07 AM, Chrysostomos Christofi <
> ch.christ...@logicom.net > wrote:
> >> For  many peoples this question maybe it’s funny , but if you know I
> >> would like to know if I can bring an electronic cigarette into the
> >> class
> >>
> >>
> >>
> >>
> >>
> >>
> >> ___
> >> 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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> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
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[OSL | CCIE_Voice] Passed

[Steffen Bruening]

I went on Thursday to Brussels for my 2nd attempt and I got my results
suprisingly today. I passed. CCIE 37891.

I like to say thank you to all on you on the list which shares your
questions and answers about the blueprint task and ipexpert labs.

I also wanna say thank you to IPexpert for their really really good
learning materials.

Regards

Steffen
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Re: [OSL | CCIE_Voice] QOS Best Pract

[Steffen Bruening]

No grading is not affected, because it is allowed to use manual qos config
and so you can use whatever you want for the names.

Am Samstag, 5. Januar 2013 schrieb Pixar Perfect :

>  Ignore the RTP header compressions part. My question is not technical but
> from strategy point. Do you think the grading could be affected if we
> remove the names built by the QOS script and use our custom names?
>
> Thanks
>
> --
> Date: Sat, 5 Jan 2013 08:55:57 +0100
> Subject: Re: [OSL | CCIE_Voice] QOS Best Pract
> From: stbruen...@gmail.com  'stbruen...@gmail.com');>
> To: pixarperf...@live.com  'pixarperf...@live.com');>
> CC: ccie_voice@onlinestudylist.com  'ccie_voice@onlinestudylist.com');>
>
> The problem with your example is that the frame-relay ip rtp
> header-compression is out-dated and the header compression should be in the
> policy-map in the rtp class.
>
> Also I don't see any less flexibilty in using auto qos, you can adjust all
> the values as with the manual way.
>
> Regards
>
> Steffen
>
> Am Samstag, 5. Januar 2013 schrieb Pixar Perfect :
>
>  What is the best practice for the real lab QOS after running the AutoQos
> - use the AutoQoS nomenclature and make changes to the classmaps,
> policymaps and map-class OR copy the AutoQOS output on a notepad and rename
> policymaps, classmaps and mapclass?
>
> I typically follow the second route as it give me more flexibility to
> "finalize" the configs on the notepad and then just copy paste instead of
> changing the AutoQOS configs on the router itself. Any inputs , comments or
> suggestions especially from those who had success with the second approach?
>
> Example:
>
> !
>
> class-map match-any RTP
>
>  match ip dscp ef
>
> class-map match-any CONTROL
>
>  match ip dscp cs3
>
> !
>
> policy-map VOIP
>
>  class RTP
>
> priority 24
>
>  class CONTROL
>
> bandwidth 19
>
>  class class-default
>
> fair-queue
>
> !
>
>
> !
>
> interface Serial0/1/1:0.1 point-to-point
>
>  bandwidth 384
>
>   frame-relay interface-dlci 201
>
>   class FRVOIP
>
>  frame-relay ip rtp header-compression
>
> !
>
>
> !
>
> !
>
> map-class frame-relay FRVOIP
>
>  frame-relay cir 364800
>
>  frame-relay bc 3648
>
>  frame-relay be 0
>
>  frame-relay mincir 364800
>
>  frame-relay fragment 480
>
>  service-policy output VOIP
>
> !
>
>
>
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Re: [OSL | CCIE_Voice] QOS Best Pract

[Steffen Bruening]

The problem with your example is that the frame-relay ip rtp
header-compression is out-dated and the header compression should be in the
policy-map in the rtp class.

Also I don't see any less flexibilty in using auto qos, you can adjust all
the values as with the manual way.

Regards

Steffen

Am Samstag, 5. Januar 2013 schrieb Pixar Perfect :

>  What is the best practice for the real lab QOS after running the AutoQos
> - use the AutoQoS nomenclature and make changes to the classmaps,
> policymaps and map-class OR copy the AutoQOS output on a notepad and rename
> policymaps, classmaps and mapclass?
>
> I typically follow the second route as it give me more flexibility to
> "finalize" the configs on the notepad and then just copy paste instead of
> changing the AutoQOS configs on the router itself. Any inputs , comments or
> suggestions especially from those who had success with the second approach?
>
> Example:
>
> !
>
> class-map match-any RTP
>
>  match ip dscp ef
>
> class-map match-any CONTROL
>
>  match ip dscp cs3
>
> !
>
> policy-map VOIP
>
>  class RTP
>
> priority 24
>
>  class CONTROL
>
> bandwidth 19
>
>  class class-default
>
> fair-queue
>
> !
>
>
> !
>
> interface Serial0/1/1:0.1 point-to-point
>
>  bandwidth 384
>
>   frame-relay interface-dlci 201
>
>   class FRVOIP
>
>  frame-relay ip rtp header-compression
>
> !
>
>
> !
>
> !
>
> map-class frame-relay FRVOIP
>
>  frame-relay cir 364800
>
>  frame-relay bc 3648
>
>  frame-relay be 0
>
>  frame-relay mincir 364800
>
>  frame-relay fragment 480
>
>  service-policy output VOIP
>
> !
>
>
>
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Re: [OSL | CCIE_Voice] GK to CUE Direct Call

[Steffen Bruening]

Did you see the call incoming on the cme? debug voip dial-peer?

Am Freitag, 4. Januar 2013 schrieb Cory Gray :

> All,
>
> ** **
>
> I noticed during the following setup that I cannot call DIRECTLY into CUE
> but if SiteC-Phone1 is busy or does not answer, forwarding the voicemail
> works.  I am not even sure you would lose points for this but I always like
> to make sure things like this work.
>
> ** **
>
> Setup is as follows.
>
> ** **
>
> CUCM > Site A GK > Site C Gateway > CUE
>
> ** **
>
> Does anyone know what extra configuration is need to accomplish this?
>
> ** **
>
> Again, this Setup works fine
>
> ** **
>
> CUCM > Site A GK > Site C Gateway > SiteC Phone 1 >CUE
>
> ** **
>
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Re: [OSL | CCIE_Voice] Terminal used for lab is putty or CRT or any other terminal

[Steffen Bruening]

There is a trial version for 30 days.



Am Sonntag, 30. Dezember 2012 schrieb singh :

> hi Guys,
>
> Thanks for your inputs.
>
> I generally use command prompt and putty to access my routers and servers
> hence I am not sure how different is secure crt from these terminals.
>
> Besides this I see it is a paid software ( secure crt) unlike putty (
> which is free) .
>
> 1) How different is secure crt from other terminals in operation (
> keyboard shortcuts , and other tabs)?
>
> 2)Do we need to practice with secure crt as well to prevent problems that
> may arise during the exam ?
>
> The expenses involved in studying for ccie program seem to just be going
> higher and higher by the day : - )
>
>
> -singh
>
>
>
>
> -- Original message --
> From:"Steffen Bruening"< stbruen...@gmail.com  'stbruen...@gmail.com');> >
> Date: 30 Dec 12 00:05:55
> Subject: Re: [OSL | CCIE_Voice] Terminal used for lab is putty or CRT or
> any other terminal
> To: Marko Milivojevic  'mar...@ipexpert.com');>>
> Cc: ; OSL Voice **; "ccie_voice-requ...@onlinestudylist.com 'cvml', 'ccie_voice-requ...@onlinestudylist.com');>"
> **
>
> I can confirm it is SecureCRT, but an old version without tabs. I training
> with desktop shortcuts to the device connections which opens in seperate
> windows.
>
>
> 2012/12/29 Marko Milivojevic  'cvml', 'mar...@ipexpert.com');>>
>
>> I *think* Voice lab is still using SecureCRT. Does it matter whether
>> it's SSH or Telnet? :-)
>>
>> --
>> Marko Milivojevic - CCIE #18427 (SP R&S)
>> Senior CCIE Instructor - IPexpert
>>
>> On Sat, Dec 29, 2012 at 9:24 AM, singh > 'cvml', 'singh8...@in.com');>>
>> wrote:
>> >
>> > hi Guys,
>> >
>> > I am interesting in knowning the following ...
>> >
>> > 1) From the CCIE voice lab which is the terminal connnection used (
>> putty or
>> > crt)?
>> >
>> > 2)Is it ssh or telnet?
>> >
>> >
>> > -singh
>> >
>> >
>> > Get Yourself a cool, short @in.com Email ID now!
>> >
>> > ___
>> > 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
>>
>
> 
>
>
>
> Get Yourself a cool, short *@in.com* Email ID 
> now!<http://www3.in.com/sso/commonregister.php?ref=IN&utm_source=invite&utm_medium=outgoing>
>
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Re: [OSL | CCIE_Voice] Cisco Unity Connection - VM pilot and VM profile made default but not applied automatically on phones

[Steffen Bruening]

Maybe it is a good idea to use just the default profile for all the phones
which have CUCN mailbox and only to add a new profile to the users who get
a CUE mailbox.


2012/12/29 CCIEing 

> Hi virajith,
>
> You may update all your phone settings using bulk edit.
>
> On Sat, Dec 29, 2012 at 8:36 PM, virajith  wrote:
>
>> hi All,
>>
>> I am noticing that for my cisco unity connection integrated  with CUCM  -
>> the VM pilot and profile that I have made as default VM pilot and profile
>> is not getting automatically applied on my phones and therefore any call
>> forward busy or no answer gets a busy tone.
>>
>> I have to manually update the VM profile on the DN level of the phone to
>> indicate that the default
>> is the unity connection.
>>
>> How do I correct this behaviour so that the phones automatically can get
>> the default VM settings without me having to specify it?
>>
>> -Vir
>>
>>
>> 
>> Catch India as it happens with the *Rediff News App*. To download it for
>> FREE, click 
>> here
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com
>>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] Terminal used for lab is putty or CRT or any other terminal

[Steffen Bruening]

I can confirm it is SecureCRT, but an old version without tabs. I training
with desktop shortcuts to the device connections which opens in seperate
windows.


2012/12/29 Marko Milivojevic 

> I *think* Voice lab is still using SecureCRT. Does it matter whether
> it's SSH or Telnet? :-)
>
> --
> Marko Milivojevic - CCIE #18427 (SP R&S)
> Senior CCIE Instructor - IPexpert
>
> On Sat, Dec 29, 2012 at 9:24 AM, singh  wrote:
> >
> > hi Guys,
> >
> > I am interesting in knowning the following ...
> >
> > 1) From the CCIE voice lab which is the terminal connnection used (
> putty or
> > crt)?
> >
> > 2)Is it ssh or telnet?
> >
> >
> > -singh
> >
> >
> > Get Yourself a cool, short @in.com Email ID now!
> >
> > ___
> > For more information regarding industry leading CCIE Lab training, please
> > visit www.ipexpert.com
> >
> > Are you a CCNP or CCIE and looking for a job? Check out
> > www.PlatinumPlacement.com
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] Long distance calls should include 1 or not ?

[Steffen Bruening]

When they ask for 10 digits you have to send only the last ten digits. your
RP looks good, on your dial-peer you don't need the forward digits command,
because 9 and 1 are the only digits which are clearly seen they will be
stripped automatically

Am Samstag, 29. Dezember 2012 schrieb sanity insanity :

>
> hi Guys ,
>
> For long distance calls I use the following patterns...
>
>
> CUCM :-
>
> 91.[2-9]XX[2-9]XX predot strip
>
>
>
> H323 gateway :-
>
> dial-peer voice 10 pots
>  description srst 10 digit LD
>  destination-pattern 91[2-9]..[2-9]..
>  port 0/0/0:23
>  forward-digits 10
>
>
>
> Questions :
>
> 1) The requirement is that 10 digits need to be sent to PSTN.  Does this
> require the 1  to be included when sent to pstn or is
> it just [2-9]..[2-9]..  as per NANP?
>
> 2) Is my understanding correct . I see some srnds in which the 1 is also
> sent to PSTN.
>
>
>
> Regards,
> MJ
>
>
>
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Re: [OSL | CCIE_Voice] CUE correct procedure...

[Steffen Bruening]

Hi singh,

connet to the cue, check/install license then put the module offline (conf
t, offline) when it is in offline state run "restore factory default" and
wait till it ask to press any key for reload. After the reload it will go
through the intial configuration wizard, when you completed that you can
configure the integration you ask for.

Am Mittwoch, 26. Dezember 2012 schrieb singh :

> Guys,
>
> How do I verify if a CUE module is in the factory default or not?
>
> -If the CUE module is factory default then I am thinking ...
>
> 1) to first check the license installed . If the license installed is not
> correct then will install
> the license
>
> 2) complete all config on cue via the cli
>
> 3) reload the CUE module ( please note the reload will be done only once)
>
>
> Is the steps (1-3) the correct way of doing it as reloading once would say
> time instead
> of having to do it again and again. Let me know if the points 1-3 is the
> correct procedure?
>
> -singh
>
> Get Yourself a cool, short *@in.com* Email ID 
> now!
>
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Re: [OSL | CCIE_Voice] MVA

[Steffen Bruening]

Only in the connect state, not in the ringing state

Am Sonntag, 23. Dezember 2012 schrieb Rrcrumm :

> Sorry
> I call from the local site B PSTN phone that is configured for a RDP and RD
>
> I call the MVA number put in my pin, press 1 to dial a number and dial
> 2001.
>
> When I look at the screen of hqph1 I just see the extension 3001 is
> calling, no name.
>
> Should I see a name and number?
>
> Thanks
> Randall
>
> On Dec 22, 2012, at 10:36 PM, Steffen Bruening 
> >
> wrote:
>
> What so you want to see? You are calling from a PSTN number into the XML
> Script. When you call a internal number through the mva, you will the
> number am when the destination hooks up and the call is connected you will
> also see the name.
>
> Am Sonntag, 23. Dezember 2012 schrieb Rrcrumm :
>
>> Hello
>> I have set up MVA and it seems to be working properly
>>
>> One question,
>> Are we suppose to see name and number when we make a call to the MVA
>> number and then press one to dial another number and we cal an internal
>> extension?
>>
>> Thanks
>> Randall
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com
>>
>
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Re: [OSL | CCIE_Voice] MVA

[Steffen Bruening]

What so you want to see? You are calling from a PSTN number into the XML
Script. When you call a internal number through the mva, you will the
number am when the destination hooks up and the call is connected you will
also see the name.

Am Sonntag, 23. Dezember 2012 schrieb Rrcrumm :

> Hello
> I have set up MVA and it seems to be working properly
>
> One question,
> Are we suppose to see name and number when we make a call to the MVA
> number and then press one to dial another number and we cal an internal
> extension?
>
> Thanks
> Randall
> ___
> 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 gateway : DDI for International Route Pattern options

[Steffen Bruening]

Why is predot trailing needed for mgcp? I use always only predot and it
works perfect.


2012/12/22 Chrysostomos Christofi 

>  Ramcharan , hi
>
> ** **
>
> Both route patterns for the below are correct 
>
> It depends what the question ask , or what is your requirement
>
> ** **
>
> **1)  **With #  in the h323 you don’t have to use Predot trailing # you 
> can use only Predot
> but for MGCP GW its need it. But its not any harm if you use  Predot
> trailing # for h323 also
>
> Regards
>
> ** **
>
> ** **
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Ramcharan Arya
> *Sent:* Σάββατο, 22 Δεκεμβρίου 2012 8:14 μμ
>
> *To:* ccie_voice@onlinestudylist.com
> *Subject:* [OSL | CCIE_Voice] MGCP gateway : DDI for International Route
> Pattern options
>
> ** **
>
> Hello,
>
>
> I have a requirement to use SLRG for International destination:
>
> First gateway : Site B ( H.323)
> Second gateway : Site A (MGCP)
>
>
> My route patterns are below:
>
> RP : 9011.!  pt: pt-sa-int
> RP : 9011.i#  pt: pt-sa-int
>
> Option : 1
> RL: SLRG
> DDI: NANP: Predot
>
> Option : 2
> RL: SLRG
> DDI: NANP: Predot trailing #
>
>
> Can you suggest which DDI is correct  correct for the above route pattern.
> Thanks & Regards,
> Ramcharan Arya
> CCIE # 28926 (R&S)
>
> 
>
>
> ___
> 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] ProctorLab Phones 7960/7962

[Steffen Bruening]

7962s are not guranteed. HQ SW is connected with one 7960 and one 7962,
site B also. Only site C has two 7962s at most of the PODs. The problem is
that for SiteB and C the phones not always connected directly, sometime
they are remotly connect via vpn, therefore the port is up, but cdp shows
nothing.

I am happy that I can mix up with my own phones, otherwise some sessions
are not very funny with only two physical phones and I don't like this ip
blue and x-lite stuff.

Am Freitag, 21. Dezember 2012 schrieb Mohamed Gazzaz :

>  Sometimes the phones are connected to different ports. Try to un-shut the
> other ports and see if they show up.
>
> Regards,
> Mohamed Gazzaz
>
> --
> Date: Fri, 21 Dec 2012 14:18:34 +
> From: sp1...@yahoo.co.uk  'sp1...@yahoo.co.uk');>
> To: ccie_voice@onlinestudylist.com  'ccie_voice@onlinestudylist.com');>
> Subject: [OSL | CCIE_Voice] ProctorLab Phones 7960/7962
>
>  Q. Do all the PL voice racks actually have  7962 phones connected to
> the sw ports  ?.
>
> I have been using the racks for some and know about the various
> 'mis-configs' inserted for troubleshooting  - but when you apply the same
> phone config for 7960 phones ports and these phones appear  in sh cdp and
> the 7962 don't - it makes you wonder  if the 7962 phone is actually
> connected.
>
> Normally I have my hardware phones on via a remote vpn link  , but on
> this session  i don't and and trying to use pview. How do you do the OWLE
> Labs using + dialling when you only see 7960 phones on the PL racks ?
> Unfortunately this is not the first time I have come across missing 7962
> and have also heard this mentioned by some attendees on a recent IP expert
> bootcamp.
>
>  So what do you do to verify whether the rack has 7962's ?
>
>  I guess IPEx/PL don't guarentee every rack has 7962 phones as when on a
> bootcamp they had to tweak the configs [using ILMED ?] to ensure the the
> candidates  actually had the 7962 phones on their PODS  they could do the
> labs.
>
> I have emailed the support team to verify but they have not/do not reply.
> Seems i'm screwed on this session.
>
> Any opinions ?
>
> .Sanjay
>
> ___ For more information
> regarding industry leading CCIE Lab training, please visit
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Re: [OSL | CCIE_Voice] Documents and other details

[Steffen Bruening]

Hi,

you need one identy document, passport for example. Lunch is included in
the exam fee, you will get a coupon for one meal, salat, drink and
dessert. In Brussels waa also a machine near the exam room were you get
water and tea for free during the exam.

Regards

Steffen

Am Donnerstag, 20. Dezember 2012 schrieb virajith :

> hi Guys,
>
> I have a question regarding if we need to carry any identity documents
> when we go to the  exam center such as
> - driver's license or passport photocopy .
> - Also do we need to carry lunch and water or is  it provided?
> - any other details  or documents we need to carry with us when we go to
> the exam center for taking the exam?
> - Is it a desktop they give us to operate? Does it have 2 monitors?
>
> Thanks,
> Vir
>
>
>
>
> 
> Catch India as it happens with the *Rediff News App*. To download it for
> FREE, click 
> here
>
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Re: [OSL | CCIE_Voice] VOL1 : Lab 9 Task 2: BACD Problem

[Steffen Bruening]

Hi,

I would always go with a pots dial-peer.

Your external BACD dial in number is truncated to +343500   and therefore
it could not find a matching dial-peer, check your voice translation.

Am Montag, 17. Dezember 2012 schrieb Ramcharan Arya :

> Hello,
>
>  I have configured BACD on Brach2 site CME When I dial pilot number  3500
> from PSTN Line 4 : 3214-1891 It generate reorder tone and call is coming on
> router and matching outbound dial-peer but script is not working.
>
> Below is my configuration and debug output.
>
>
>  service queue flash:bacdprompts/app-b-acd-2.1.2.2.tcl
>   param queue-len 15
>   param aa-hunt10 3006
>   param queue-manager-debugs 1
>   param aa-hunt2 3210
>   param number-of-hunt-grps 2
>  !
>  service aa flash:bacdprompts/app-b-acd-aa-2.1.2.2.tcl
>   paramspace english index 1
>   param number-of-hunt-grps 2
>   param handoff-string aa
>   paramspace english language en
>   param max-time-vm-retry 2
>   param aa-pilot 3500
>   paramspace english location flash:bacdprompts/
>   param second-greeting-time 60
>   param welcome-prompt _bacd_welcome.au
>   param call-retry-timer 15
>   param voice-mail 3001
>   param max-time-call-retry 90
>   param service-name queue
> !
> dial-peer voice 222 voip
>  service aa
>  destination-pattern 3500
>  session target ipv4:10.10.110.3
>  incoming called-number 3500
>  dtmf-relay h245-alphanumeric
>  codec g711ulaw
>  no vad
>
> ---
>
>
> Dec 17 16:27:23.328: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8  callref =
> 0x00D3
> Bearer Capability i = 0x8090A3
> Standard = CCITT
> Transfer Capability = Speech
> Transfer Mode = Circuit
> Transfer Rate = 64 kbit/s
> Channel ID i = 0xA98381
> Exclusive, Channel 1
> Progress Ind i = 0x8583 - Origination address is non-ISDN
> Calling Party Number i = 0x2180, '32141891'
> Plan:ISDN, Type:National
> Called Party Number i = 0x80, '+3432143500'
> Plan:Unknown, Type:Unknown
> Dec 17 16:27:23.328: ISDN Se0/0/0:15 Q931: Received SETUP  callref =
> 0x80D3 callID = 0x0032 switch = primary-net5 interface = User
> Dec 17 16:27:23.332: //-1//DPM/dpAssociateIncomingPeerCore:
>Calling Number=32141891, Called Number=+343500,
> Voice-Interface=0x49FFDE10,
>Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search
> Type=PEER_TYPE_VOICE,
>Peer Info Type=DIALPEER_INFO_SPEECH
> Dec 17 16:27:23.332: //-1//DPM/dpAssociateIncomingPeerCore:
>Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1
> Dec 17 16:27:23.332: //-1//DPM/dpAssociateIncomingPeerCore:
>Calling Number=32141891, Called Number=+343500, Voice-Interface=0x0,
>Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search
> Type=PEER_TYPE_VOICE,
>Peer Info Type=DIALPEER_INFO_FAX
> Dec 17 16:27:23.332: //-1//DPM/dpAssociateIncomingPeerCore:
>Result=NO_MATCH(-1) After All Match Rules Attempt
> Dec 17 16:27:23.332: //-1//DPM/dpMatchPeersCore:
>Calling Number=, Called Number=+343500, Peer Info
> Type=DIALPEER_INFO_SPEECH
> Dec 17 16:27:23.332: //-1//DPM/dpMatchPeersCore:
>Match Rule=DP_MATCH_DEST; Called Number=+343500
> Dec 17 16:27:23.332: //-1//DPM/dpMatchPeersCore:
>No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)
> Dec 17 16:27:23.332: //-1//DPM/dpMatchPeers:
>Result=NO_MATCH(-1)
> Dec 17 16:27:23.340: //-1/78994D148033/DPM/dpMatchPeersCore:
>Calling Number=, Called Number=+343500, Peer Info
> Type=DIALPEER_INFO_SPEECH
> Dec 17 16:27:23.340: //-1/78994D148033/DPM/dpMatchPeersCore:
>Match Rule=DP_MATCH_DEST; Called Number=+343500
> Dec 17 16:27:23.340: //-1/78994D148033/DPM/dpMatchPeersCore:
>No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)
> Dec 17 16:27:23.340: //-1/78994D148033/DPM/dpMatchPeersMoreArg:
>Result=NO_MATCH(-1)
> Dec 17 16:27:23.340: //-1//DPM/dpMatchPeersCore:
>Calling Number=+343500, Called Number=+343500, Peer Info
> Type=DIALPEER_INFO_SPEECH
> Dec 17 16:27:23.340: //-1//DPM/dpMatchPeersCore:
>Match Rule=DP_MATCH_DEST; Called Number=+343500
> Dec 17 16:27:23.340: //-1//DPM/dpMatchPeersCore:
>No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)
> Dec 17 16:27:23.340: //-1//DPM/dpMatchPeersMoreArg:
>Result=NO_MATCH(-1)
> Dec 17 16:27:23.340: //-1//DPM/dpMatchPeersCore:
>Calling Number=+343500, Called Number=+343500, Peer Info
> Type=DIALPEER_INFO_SPEECH
> Dec 17 16:27:23.340: //-1//DPM/dpMatchPeersCore:
>Match Rule=DP_MATCH_DEST; Called Number=+343500
> Dec 17 16:27:23.340: //-1//DPM/dpMatchPeersCore:
>No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)
> Dec 17 16:27:23.340: //-1//DPM/dpMatchPeersMoreArg:
>Result=NO_MATCH(-1)
> Dec 17 16:27:23.340: //-1//DPM/dpAssociateIncomingPeerCore:

Re: [OSL | CCIE_Voice] CUCM TFTP service restart from CLI

[Steffen Bruening]

Thats not allowed via CLI, so no syntax available.

Am Montag, 3. Dezember 2012 schrieb Ramcharan Arya :

> Hello,
>
> Can you someone please tell me command syntax how to restart Cisco Tftp
> service from CUCM CLI.?
>
> Thanks & Regards,
> Ramcharan Arya
> CCIE # 28926 (R&S)
>
>
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Re: [OSL | CCIE_Voice] Interface for binding?

[Steffen Bruening]

You don't have to change the routing, routing is done for you. You can the
reach the CUCM from any interface.

Am Sonntag, 2. Dezember 2012 schrieb virajith :

> hi Steffen,
>
> Thanks for your reply
>
>
> If we can bind to any interface ( gig , loopback ) then is it necessary
> for us to setup the the  routing as well as  traffic from loopback or
> virtual interfaces needs to reach the callmanger and other devices also if
> the routing needs to setup then this is going to eat into the allocated
> exam time?
>
>
> -Vir
>
>
>
>
> From: Steffen Bruening  'stbruen...@gmail.com');>>
> Sent: Sat, 01 Dec 2012 12:24:35
> To: virajith  'vir...@rediffmail.com');>>
> Cc: "ccie_voice@onlinestudylist.com  'ccie_voice@onlinestudylist.com');>" 
>  'ccie_voice@onlinestudylist.com');>
> >
> Subject: Re: [OSL | CCIE_Voice] Interface for binding?
> Generally it doesn't matter to which you bind your media resources. But
> maybe the question ask you for a specific binding.
>
> The only thing you never should do is to bind Cube and Gatekeeper to the
> same Interface.
>
> Regards
>
> Steffen
>
> Am Samstag, 1. Dezember 2012 schrieb virajith :
>
> Hi All,
>
> I am wondering which would be the best interface to bind for transcoders ,
> gateways , cfbs and for sccp configurations I know loopback is the best for
> binding but with like to clarify with the example below...
>
> For example...
>
> R3#sh ip int brief
> Interface   IP-Address   OK? Method Status Protocol
> Serial0/0/1:0   unassigned   YES NVRAM up up
> Serial0/0/1:0.101  142.1.67.2 YES NVRAM up up
> Vlan1unassigned   YES NVRAM up up
> Vlan502   142.102.66.254   YES NVRAM up up
> Vlan602   142.202.66.254   YES NVRAM up up
> SSLVPN-VIF0   unassigned   NO unset up up
> Loopback0 142.1.66.254   YES NVRAM up up
>
>
> In the above scenario  interface vlan502 is the is in the  voice vlan
> interface with the ip address  of  142.102.66.254  . However I have the
> loopback ip address  is 142.1.66.254.
>
> My Servers ( CUCM , CUC, CUPS, CCX) are in  142.100.64.x
>
> In this situation which is the best interface to bin with and why  ?
>
>
> -Vir
>
>
> <http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm@Middle?>
> Catch India as it happens with the *Rediff News App*. To download it for
> FREE, click 
> here<http://track.rediff.com/click?url=___http://www.rediff.com/newsapp___&lnk=signature&newservice=newsapp>
>
>
>
> <http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm@Middle?>
> Catch India as it happens with the *Rediff News App*. To download it for
> FREE, click 
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>
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Re: [OSL | CCIE_Voice] Interface for binding?

[Steffen Bruening]

Generally it doesn't matter to which you bind your media resources. But
maybe the question ask you for a specific binding.

The only thing you never should do is to bind Cube and Gatekeeper to the
same Interface.

Regards

Steffen

Am Samstag, 1. Dezember 2012 schrieb virajith :

> Hi All,
>
> I am wondering which would be the best interface to bind for transcoders ,
> gateways , cfbs and for sccp configurations I know loopback is the best for
> binding but with like to clarify with the example below...
>
> For example...
>
> R3#sh ip int brief
> Interface   IP-Address   OK? Method Status Protocol
> Serial0/0/1:0   unassigned   YES NVRAM up up
> Serial0/0/1:0.101  142.1.67.2 YES NVRAM up up
> Vlan1unassigned   YES NVRAM up up
> Vlan502   142.102.66.254   YES NVRAM up up
> Vlan602   142.202.66.254   YES NVRAM up up
> SSLVPN-VIF0   unassigned   NO unset up up
> Loopback0 142.1.66.254   YES NVRAM up up
>
>
> In the above scenario  interface vlan502 is the is in the  voice vlan
> interface with the ip address  of  142.102.66.254  . However I have the
> loopback ip address  is 142.1.66.254.
>
> My Servers ( CUCM , CUC, CUPS, CCX) are in  142.100.64.x
>
> In this situation which is the best interface to bin with and why  ?
>
>
> -Vir
>
>
> 
> Catch India as it happens with the *Rediff News App*. To download it for
> FREE, click 
> here
>
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Re: [OSL | CCIE_Voice] UCCX Prompt Question

[Steffen Bruening]

Hi Bill,

I had my first attempt in November, but failed because I don't read
some question corectly
in every detail.

I had this problem with -1 in some of my lab sessions (not proctor
labs) and it worked with contacts waiting for me. I tested it with 3
call-in users and all get an indivudually annoucement for the amount of
calls ahead.

But as I wrote, it doesn't matter how the script looks like as long as it
fit the question in your binder.

Regards

Steffen

Am Freitag, 30. November 2012 schrieb Bill Lake :

> Sounds like either way works, so as long as it works, it really doesn't
> matter how you get it, just as long as the proctor gets what he is supposed
> to hear.
>
> I know I have used the slightly longer way as it helps me stay organized
> but next lab session I am going to try this if I can keep my ducks in a row
> :)
>
> I also know from my practice that it does not say -1 when I set this up
> using Position in queue, it says your position in queue is zero or whatever
> I record.
>
> I also practice recording in both in CUC and UCCX just in case one does
> not work. You know both methods so just practice because you never know :)
>
> It seems like you might be getting close to your lab, do you have one
> schedule?
>
> Bill
>
> On Fri, Nov 30, 2012 at 4:05 PM, William Bell 
> 
> > wrote:
>
>> I have a question that may really just come down to a matter of
>> preference. However, I want to make sure there is something that I am not
>> missing. For those who want to "read along" my question stems from
>> IPexpert's One Week Lab Experience lab 2. I also think i have seen a
>> similar question in the 5-lab workbook.
>>
>> For everyone else, the CCX requirement is to play a contact's position in
>> queue while they are in the queued branch of Select Resource.
>>  Specifically, they want you to play a prompt that says "The number of
>> people ahead of you is one" (or two, or three, etc.).
>>
>> The way I do this is as follows:
>>
>> step: Select Resource from CSQ
>> - (Connected)
>> - (Queued)
>> label: queueLoop
>> intPosInQ = Get Reporting Statistic PositionInQueue
>> decrement intPosInQ
>> playPrompt (P[YourPosinQ.wav] + intPosInQ)
>> delay 30s
>> goto label: queueLoop
>>
>>
>> The way I have seen IPExpert handle this has a few more steps:
>>
>> step: Select Resource from CSQ
>> - (Connected)
>> - (Queued)
>> label: queueLoop
>> intPosInQ = Get Reporting Statistic PositionInQueue
>> decrement intPosInQ
>> promptNumInQ = Create Generated Prompt number (intPosInQ)
>> promptQueue = Create Container Prompt Concatenation (P[YourPosinQ] +
>> promptNumInQ)
>> playPrompt (promptQueue)
>> delay 30s
>> goto label: queueLoop
>>
>>
>> When I use my method, I get the desired result. My question is what (if
>> any) advantage is there in generating the spoken prompt and packaging the
>> two prompts instead of just doing the concatenation in-line with the Play
>> Prompt step?
>>
>> Thanks in advance.
>>
>> -Bill
>>  --
>> William Bell
>> blog: http://ucguerrilla.com
>> twitter: @ucguerrilla
>>
>>
>>
>> ___
>> 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] UCCX Prompt Question

[Steffen Bruening]

Hi Bill,

When you ask for callers ahead of you, you should go with
ContactsWaiting+decrement step, because position in Queue is your position
and when nobody else is in the queue you will hear a "- 1" after the
decrement, don't know whether there is also an increment step.

At the end it is your choice how you get your script working. Not the way,
only the result is graded.

Regards

Steffen

Am Freitag, 30. November 2012 schrieb William Bell :

> I have a question that may really just come down to a matter of
> preference. However, I want to make sure there is something that I am not
> missing. For those who want to "read along" my question stems from
> IPexpert's One Week Lab Experience lab 2. I also think i have seen a
> similar question in the 5-lab workbook.
>
> For everyone else, the CCX requirement is to play a contact's position in
> queue while they are in the queued branch of Select Resource.
>  Specifically, they want you to play a prompt that says "The number of
> people ahead of you is one" (or two, or three, etc.).
>
> The way I do this is as follows:
>
> step: Select Resource from CSQ
> - (Connected)
> - (Queued)
> label: queueLoop
> intPosInQ = Get Reporting Statistic PositionInQueue
> decrement intPosInQ
> playPrompt (P[YourPosinQ.wav] + intPosInQ)
> delay 30s
> goto label: queueLoop
>
>
> The way I have seen IPExpert handle this has a few more steps:
>
> step: Select Resource from CSQ
> - (Connected)
> - (Queued)
> label: queueLoop
> intPosInQ = Get Reporting Statistic PositionInQueue
> decrement intPosInQ
> promptNumInQ = Create Generated Prompt number (intPosInQ)
> promptQueue = Create Container Prompt Concatenation (P[YourPosinQ] +
> promptNumInQ)
> playPrompt (promptQueue)
> delay 30s
> goto label: queueLoop
>
>
> When I use my method, I get the desired result. My question is what (if
> any) advantage is there in generating the spoken prompt and packaging the
> two prompts instead of just doing the concatenation in-line with the Play
> Prompt step?
>
> Thanks in advance.
>
> -Bill
> --
> William Bell
> blog: http://ucguerrilla.com
> twitter: @ucguerrilla
>
>
>
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[OSL | CCIE_Voice] Learn and Motivation Strategy for the second attempt

[Steffen Bruening]

Hi all,

I was in Bruessels yesterday and I failed. I was suprised about my score
report and about the amount of section were lost points. I only know one
definitly mistake, about the other things I could only suppose.

It seems that I am just to stupid to read the questions. I was finished
with the configuration after 5 1/2 hours and I tested everything twice,
solved the issues I found, but unfortunately it was not enough.

I booked my second attempt for January, but to be honest I am really
frustated and I don't know how should keep my motivation and focus. Sure I
will make further remote labs to keep my speed, but what else could I do
till my second attempt?

So all second/more attempter what have you made during your till your next
lab seat? Did you changed or strategy/learning focus or anything else?

Regards

Steffen
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Re: [OSL | CCIE_Voice] ESW and switch config

[Steffen Bruening]

I still need the sw access mode, because there is no "spanning-tree
portfast trunk" command for the ESW and so I would lose the marks for the
vlan section because in most cases it is required to bypass all the
spanning-tree states or just that the phones should boot up as fast as
possible.

Regards
Steffen

2012/11/1 Steffen Bruening 

> Oh okay. I apologize for the confusion I made.
>
>
> 2012/11/1 Cory Gray 
>
>>  IPexpert and myself have run into problems doing it the traditional
>> way.  But for the authoritative source you should be looking at the LAN
>> Switching Guide for IOS 12.4T which confirms that recommendation.  The
>> guide you point to is for Metro 3750 switches.  The ISR routers in the lab
>> run 12.4T.
>>
>>
>> http://www.cisco.com/en/US/docs/ios-xml/ios/lanswitch/configuration/12-4t/lsw-hwic-ethsw-ic.html#GUID-379450C0-5434-4AC3-9BED-396CB7D162C1
>>
>>
>>
>> --
>> Date: Thu, 1 Nov 2012 08:08:36 +
>> Subject: Re: [OSL | CCIE_Voice] ESW and switch config
>> From: stbruen...@gmail.com
>> To: corygray22...@hotmail.com
>> CC: vir...@rediffmail.com; ccie_voice@onlinestudylist.com
>>
>>
>> Recommed by whom? Ipexpert, I know, but the problem is that the
>> Proctors/Scripts which marks your lab are not from IPExpert they are from
>> Cisco. Therefore it should be better to follow the cisco guidelines:
>>
>> "You should configure voice VLAN on switch access ports; voice VLAN is
>> not supported on trunk ports. You can only configure a voice VLAN on Layer
>> 2 ports."
>>
>>
>> http://www.ciscosystems.com/en/US/docs/switches/metro/catalyst3750m/software/release/12.2_25_ey/configuration/guide/swvoip.html#wp1030825
>>
>>
>> 2012/11/1 Cory Gray 
>>
>> You should be fine without one just know that the recommended ESW config
>> is
>>
>> Switchport mode trunk
>>
>> Switch trunk native vlan X (X equals data vlan)
>>
>> Switch voice vlan Y (y equals voice vlan)
>>
>> ** **
>>
>> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
>> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *virajith
>> *Sent:* Wednesday, October 31, 2012 11:22 PM
>> *To:* ccie_voice@onlinestudylist.com
>> *Subject:* [OSL | CCIE_Voice] ESW and switch config
>>
>> ** **
>>
>> hi Guys,
>>
>> I am just wondering if one can practice the labs without a ESW module .
>>
>> I have a setup in with I am using a switch (3750) for the vlan config.
>>
>> Is an ESW module necessary for the lab practice?
>>
>> How is the above config on 3750 different from using an ESW module?
>>
>>
>> -Vir
>>
>>
>> <http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm%40Middle?>
>> 
>>
>> Catch India as it happens with the *Rediff News App*. To download it for
>> FREE, click 
>> here<http://track.rediff.com/click?url=___http://www.rediff.com/newsapp___&lnk=signature&newservice=newsapp>
>> 
>>
>> ___
>> 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] ESW and switch config

[Steffen Bruening]

Oh okay. I apologize for the confusion I made.

2012/11/1 Cory Gray 

>  IPexpert and myself have run into problems doing it the traditional way.
> But for the authoritative source you should be looking at the LAN Switching
> Guide for IOS 12.4T which confirms that recommendation.  The guide you
> point to is for Metro 3750 switches.  The ISR routers in the lab run 12.4T.
>
>
> http://www.cisco.com/en/US/docs/ios-xml/ios/lanswitch/configuration/12-4t/lsw-hwic-ethsw-ic.html#GUID-379450C0-5434-4AC3-9BED-396CB7D162C1
>
>
>
> --
> Date: Thu, 1 Nov 2012 08:08:36 +
> Subject: Re: [OSL | CCIE_Voice] ESW and switch config
> From: stbruen...@gmail.com
> To: corygray22...@hotmail.com
> CC: vir...@rediffmail.com; ccie_voice@onlinestudylist.com
>
>
> Recommed by whom? Ipexpert, I know, but the problem is that the
> Proctors/Scripts which marks your lab are not from IPExpert they are from
> Cisco. Therefore it should be better to follow the cisco guidelines:
>
> "You should configure voice VLAN on switch access ports; voice VLAN is
> not supported on trunk ports. You can only configure a voice VLAN on Layer
> 2 ports."
>
>
> http://www.ciscosystems.com/en/US/docs/switches/metro/catalyst3750m/software/release/12.2_25_ey/configuration/guide/swvoip.html#wp1030825
>
>
> 2012/11/1 Cory Gray 
>
> You should be fine without one just know that the recommended ESW config is
> 
>
> Switchport mode trunk
>
> Switch trunk native vlan X (X equals data vlan)
>
> Switch voice vlan Y (y equals voice vlan)
>
> ** **
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *virajith
> *Sent:* Wednesday, October 31, 2012 11:22 PM
> *To:* ccie_voice@onlinestudylist.com
> *Subject:* [OSL | CCIE_Voice] ESW and switch config
>
> ** **
>
> hi Guys,
>
> I am just wondering if one can practice the labs without a ESW module .
>
> I have a setup in with I am using a switch (3750) for the vlan config.
>
> Is an ESW module necessary for the lab practice?
>
> How is the above config on 3750 different from using an ESW module?
>
>
> -Vir
>
>
> 
> 
>
> Catch India as it happens with the *Rediff News App*. To download it for
> FREE, click 
> here
> 
>
> ___
> 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] Uccx scripting making me mad!!

[Steffen Bruening]

Here you can find some scripting guides:

http://www.cisco.com/en/US/products/sw/custcosw/ps1846/products_programming_reference_guides_list.html



2012/11/1 sanity insanity 

> hi Guys,
>
> I really need your help to understand UCCX scripts ...How they are made?
> and how they work?
>
> Please help guys!
>
> -Mark
>
> ___
> 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] ESW and switch config

[Steffen Bruening]

Recommed by whom? Ipexpert, I know, but the problem is that the
Proctors/Scripts which marks your lab are not from IPExpert they are from
Cisco. Therefore it should be better to follow the cisco guidelines:

"You should configure voice VLAN on switch access ports; voice VLAN is not
supported on trunk ports. You can only configure a voice VLAN on Layer 2
ports."

http://www.ciscosystems.com/en/US/docs/switches/metro/catalyst3750m/software/release/12.2_25_ey/configuration/guide/swvoip.html#wp1030825


2012/11/1 Cory Gray 

> You should be fine without one just know that the recommended ESW config is
> 
>
> Switchport mode trunk
>
> Switch trunk native vlan X (X equals data vlan)
>
> Switch voice vlan Y (y equals voice vlan)
>
> ** **
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *virajith
> *Sent:* Wednesday, October 31, 2012 11:22 PM
> *To:* ccie_voice@onlinestudylist.com
> *Subject:* [OSL | CCIE_Voice] ESW and switch config
>
> ** **
>
> hi Guys,
>
> I am just wondering if one can practice the labs without a ESW module .
>
> I have a setup in with I am using a switch (3750) for the vlan config.
>
> Is an ESW module necessary for the lab practice?
>
> How is the above config on 3750 different from using an ESW module?
>
>
> -Vir
>
>
> 
> 
>
> Catch India as it happens with the *Rediff News App*. To download it for
> FREE, click 
> here
> 
>
> ___
> 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] did issue

[Steffen Bruening]

Q931 doesn"t lie. Something is missing in your config, maybe on the cucm,
maybe on the gateway (for h323). It depends on where your stripping your
DIDs to the internal number format.

2012/10/30 otunola Akerele 

> hi all, am new to the forum, ples am having some issues am using
> proctorlabs and following the 5lab workbook
>
> 1.i get a fast busy when i place a call to the pstn from siteA and again
> what could be the issue if i place a call from the pstn to site b and the
> debug says unallocated/unknown number whereas every of the configuration
> looks fine
>
> ___
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> visit www.ipexpert.com
>
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> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] Unable to call AVT from PSTN 911

[Steffen Bruening]

Hi,

you don't need such a bunch of dial-peers for CUE

You are using MWI outcalling, therefore you only need the both DN's also
you can send Voicemal and AVT on the same Dial-Peer to the CUE, use 400[45]
as destination-pattern.

If outcalling is not request in the question I would prefer unsolicited
notification to the CME.

Check your AVT sip trigger on the cue cli.

Regards

Steffen

2012/10/25 Ramcharan Arya 

> Hello,
>
> I am working on a task trying to call AVT extension  4005 from PSTN
> 911. I am able to call voice mail and auto-attendent but when I call
> AVT number I get re-order tone.
>
> Has anyone try AVT setup on CUE. Please let me know.
>
> I got another problem with MWI lamp is not working when use left voice
> mail to extension.
>
>
> telephony-service
> no auto-reg-ephone
>   max-ephones 52
>  max-dn 192
>  ip source-address 10.10.128.1 port 2000
>  voicemail 4004
>  max-conferences 8 gain -6
>  moh music-on-hold.au
>  multicast moh 239.23.4.10 port 2000
>  web admin system name Admin password admin
>  dn-webedit
>  time-webedit
>  transfer-system full-consult
> !
> ephone-dn  98
>  number 4010
>  mwi off
> !
> !
> ephone-dn  99
>  number 4011
>  mwi on
>
> !
> dial-peer voice 98 voip
>  description MWI OFF
>  incoming called-number 4010
>  codec g711ulaw
>  no vad
> !
> dial-peer voice 99 voip
>  description MWI ON
>  incoming called-number 4011
>  codec g711ulaw
>  no vad
> !
> dial-peer voice 4004 voip
>  destination-pattern 4004
>  session protocol sipv2
>  session target ipv4:10.10.128.2
>  dtmf-relay sip-notify
>  codec g711ulaw
>  no vad
> !
> dial-peer voice 4005 voip
>** Description AVT ***
>  destination-pattern 4005
>  session protocol sipv2
>  session target ipv4:10.10.128.2
>  dtmf-relay sip-notify
>  codec g711ulaw
>  no vad
>
> Can you please  suggest is something is missing in my configuration.
>
> Thanks & Regards,
> Ramcharan Arya
> ___
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> visit www.ipexpert.com
>
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>
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Re: [OSL | CCIE_Voice] LAN Qos questions

[Steffen Bruening]

Hi Krishna,

sounds good so far. But as I said when COS 5 is the only COS in Queue 1,
which Traffic/COS could fill T1 and T2 up to 100% when COS 5 is only mapped
to T3?

Regards

Steffen

2012/10/21 Krishna 

> steffen,
>
> your approach is not right way of doing it because when u look the
> threshold values of the queues you have allocated max threshold is 100 and
> reserved threshold is 100, guess what both threshold i.e. t1 and t2 takes
> up to 100% value when desired and that being said after t1 and t2 were
> filled it comes to t3 which has 75% i.e. it is the last threshold where it
> will take/borrow the memory value from reserved threshold when desired.
> long story short... right way of doing it either assign it to t2 or t1 and
> assign threshold value of 75% for correct approach...
>
> thank you
> krishna.
>   ------
> *From:* Steffen Bruening 
> *To:* Pixar Perfect 
> *Cc:* ccie_voice@onlinestudylist.com
> *Sent:* Saturday, October 20, 2012 6:38 PM
> *Subject:* Re: [OSL | CCIE_Voice] LAN Qos questions
>
> I have this seen this also, to be honest I think it shouldn't matter
> whether it is in threshold 1 or 3 as long as no other COS is in same
> Threshold of queue 1 of queset 2. When you leave in in threshold 3 I think
> you should be fine with:
>
> mls qos queue-set output *2* threshold 1  100* *100* 75 *100.
>
> Maybe I am completly wrong but thats they way I understood this.
>
> Regards
>
> Steffen
>
> 2012/10/20 Pixar Perfect 
>
>  The requirement is as follows on Lab 2 QOS section of the 5-Lab Handbook.
> "For traffic being sent to the Site A gateway ensure that the traffic
> marked with COS 5 is dropped if the queue 1 is 75% full"
>
> The Solution guide (page 408) has the following solution.
>
> mls qos queue-set output *2* threshold 1  *75 100 100 100*   --> queset
> is preconfigured on the port to 2
> mls qos srr-queue output cos-map queue 1 threshold *3*   5
>
> ..
> My interpretation was to move the Cos 5 into Q1t1 but the command says
> threshold 3 .. is this just a typo or am I missing something obvious.
>
>
> Thanks!
>
> ___
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> visit www.ipexpert.com
>
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>
>
>
> ___
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>
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Re: [OSL | CCIE_Voice] LAN Qos questions

[Steffen Bruening]

I have this seen this also, to be honest I think it shouldn't matter
whether it is in threshold 1 or 3 as long as no other COS is in same
Threshold of queue 1 of queset 2. When you leave in in threshold 3 I think
you should be fine with:

mls qos queue-set output *2* threshold 1  100* *100* 75 *100.

Maybe I am completly wrong but thats they way I understood this.

Regards

Steffen

2012/10/20 Pixar Perfect 

>  The requirement is as follows on Lab 2 QOS section of the 5-Lab Handbook.
> "For traffic being sent to the Site A gateway ensure that the traffic
> marked with COS 5 is dropped if the queue 1 is 75% full"
>
> The Solution guide (page 408) has the following solution.
>
> mls qos queue-set output *2* threshold 1  *75 100 100 100*   --> queset
> is preconfigured on the port to 2
> mls qos srr-queue output cos-map queue 1 threshold *3*   5
>
> ..
> My interpretation was to move the Cos 5 into Q1t1 but the command says
> threshold 3 .. is this just a typo or am I missing something obvious.
>
>
> Thanks!
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
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> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] RSVP brokes transcoding

[Steffen Bruening]

Hi Cory,

that's the trick. Thanks a lot.

2012/10/20 Cory Gray 

> Do you have g729r8 on the transcoder?  It is not their by default so you
> would have to add it to the list.  I always add it to conferencing and
> transcoder just to get in the habit.
>
> Sent from my iPhone
>
> On Oct 20, 2012, at 4:32 PM, "Steffen Bruening" 
> wrote:
>
> > Hi,
> >
> > I have 3 Sites, all of them configured on the CUCM, Site C has Voicemail
> with local CUE. When I am dialing from Site B to C codec g279 will be used
> and I can reach the voicemail, so I know When I am dialing from Site A to C
> through an RSVP CAC Location I get a fast busy when reaching the VM Pilot
> of Site C. When I take of the RSVP I can also reach VM Pilot of Site C with
> G729.
> >
> > Maybe somebody can explain to me why these RSVP Calls to VM are failing.
> >
> > My workaround for this to create a new location for VM Ports and Route
> Points which does not use RSVP. But this will brake the requirement of the
> question to allow only 4 calls between HQ und SC over the WAN.
> >
> > Regards
> >
> > Steffen
> > ___
> > 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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[OSL | CCIE_Voice] RSVP brokes transcoding

[Steffen Bruening]

Hi,

I have 3 Sites, all of them configured on the CUCM, Site C has Voicemail
with local CUE. When I am dialing from Site B to C codec g279 will be used
and I can reach the voicemail, so I know When I am dialing from Site A to C
through an RSVP CAC Location I get a fast busy when reaching the VM Pilot
of Site C. When I take of the RSVP I can also reach VM Pilot of Site C with
G729.

Maybe somebody can explain to me why these RSVP Calls to VM are failing.

My workaround for this to create a new location for VM Ports and Route
Points which does not use RSVP. But this will brake the requirement of the
question to allow only 4 calls between HQ und SC over the WAN.

Regards

Steffen
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Re: [OSL | CCIE_Voice] lan qos- cos 5 threshold 1 in queue 1 doesn't show in running config

[Steffen Bruening]

Hi Krishna,

you will not see Cos 5 in Threshold 1 because this is the default Threshold
for Cos 5. The running-config presents always only the non-default values,
e.g if you put an prefererence command on your dial-peers you will this
only as long as it is not  the default value 0.

Regards

Steffen

2012/10/14 Krishna 

> hi guys,
>
> i was wondering whether i am doing right way of doing lan qos or not ??
> the requirements are assign cos 5 to priority queue , cos 3 4  to queue 2
> with 60% exceed of cos 4 should be dropped. so here is my configuration for
> that
>
> mls qos
> mls qos srr-queue output cos-map queue 1 threshold 1 5
> mls qos srr-queue output cos-map queue 2 threshold 2 3
> mls qos srr-queue output cos-map queue 2 threshold 1 4
>
> mls qos queue-set output 2 threshold 3 60 100 100 272
>
> when i issued show run | i  mls commands, i see every  mls qos command
> except the cos 5 which is assigned to q1 t1.  Is my approach is correct in
> dealing this question correctly?? does it matter whether we assign cos
> values to t1 or t2 or t3 in the queues???
>
> your input is much appreciated.
>
> thank you
> krishna.
>
> ___
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> visit www.ipexpert.com
>
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Re: [OSL | CCIE_Voice] CME Multicast MOH Port number

[Steffen Bruening]

It's from multicast moh section of the cmeadmin guide, page 1252 of the
current guide version. I think that has nothing to do with live-feed in
this case.

2012/10/4 Dimuthu 

> For me it seems document is correct (no typos). We can configure different
> moh live-feeds with unique media port numbers using voice
> moh-group  command. Document Author seems referring to the
> live-feed with media port 2000 inside telephony-service configuration.
>
>   *From:* Jason Aarons (AM) 
> *To:* Steffen Bruening ; ccie_voice <
> ccie_voice@onlinestudylist.com>
> *Sent:* Wednesday, October 3, 2012 8:59 PM
> *Subject:* Re: [OSL | CCIE_Voice] CME Multicast MOH Port number
>
>   I think that is a typo.  Stick with 16384.
>
>  *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Steffen Bruening
> *Sent:* Tuesday, October 02, 2012 5:28 PM
> *To:* ccie_voice
> *Subject:* [OSL | CCIE_Voice] CME Multicast MOH Port number
>
>
>
> Hi all,
>
>  I know that usually we would configure multicast moh for CME in this way
> "multicast' moh 239.1.1.1 port 16384 route ". I read through the CME
> admin guide and found the following sentence:
>
>  *port port-number—Media port for multicast. Range is 2000 to 65535. We
> recommend port 2000 because it is already used for normal RTP media
> transmissions between IP phones and the router.*
>
>  I use hardware vpn with Proctorlabs so I can't test multicast moh with
> my phones at home. Did somebody uses Port 2000 for multicast moh?
>
>  Regards
>
>  Steffen
>
>
> itevomcid
>
> ___
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>
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>
>
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[OSL | CCIE_Voice] DSP Allocation

[Steffen Bruening]

Hi all,

I have a fractional PRI with MGCP (8 channels). When I look int "sh voice
dsp group all" I could see that 8 channels of dsp 5 are used. Maybe someone
can explain how the dsp allocation algorithm is working. I would like to
know why not dsp 1 will be used. Is there maybe an option that I could
decide which dsp should be used (a kind of preference)???

dsp 1:
  State: UP, firmware: 9.4.7
  Max signal/voice channel: 8/8
  Max credits: 160
  num_of_sig_chnls_allocated: -2
  Transcoding channels allocated: 0
  Group: FLEX_GROUP_VOICE, complexity: FLEX
Shared credits: 160, reserved credits: 0
Signaling channels allocated: 0
Voice channels allocated: 0
Credits used: 0

dsp 5:
  State: UP, firmware: 9.4.7
  Max signal/voice channel: 16/16
  Max credits: 240
  num_of_sig_chnls_allocated: 8
  Transcoding channels allocated: 0
  Group: FLEX_GROUP_VOICE, complexity: FLEX
Shared credits: 240, reserved credits: 0
Signaling channels allocated: 8
Voice channels allocated: 0
Credits used: 0

Regards

Steffen
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[OSL | CCIE_Voice] CME Multicast MOH Port number

[Steffen Bruening]

Hi all,

I know that usually we would configure multicast moh for CME in this way
"multicast' moh 239.1.1.1 port 16384 route ". I read through the CME
admin guide and found the following sentence:

*port port-number—Media port for multicast. Range is 2000 to 65535. We
recommend port 2000 because it is already used for normal RTP media
transmissions between IP phones and the router.*

I use hardware vpn with Proctorlabs so I can't test multicast moh with my
phones at home. Did somebody uses Port 2000 for multicast moh?

Regards

Steffen
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Re: [OSL | CCIE_Voice] sip gateway PSTN disconnect cause delayed send to cisco phone

[Steffen Bruening]

Hi Krishna,

I fixed it with ios software mtp and mtp required on the sip trunk.

2012/8/30 Krishna 

> steffen,
>
> do you have other gw listed in the route list??? 404 sip response tries to
> hunt next gw if it is listed in route list that may causing this issue...
>
> thank you
> Krishna.
>
>   ------
> *From:* Steffen Bruening 
> *To:* ccie_voice 
> *Sent:* Thursday, August 30, 2012 4:56 AM
> *Subject:* [OSL | CCIE_Voice] sip gateway PSTN disconnect cause delayed
> send to cisco phone
>
> Hi all,
>
> this question is not ccie lab related. I have 2911 integrated via sip to
> cucm 8.6. When a call terminates because of user busy/unallocated number or
> what ever, the cisco phone still rings for 30 seconds because the sip
> disconnect message was send 30 seconds after pstn disconnect. How could I
> reduce this timer?
>
> *Aug 30 09:50:55.631: ISDN Se0/0/0:15 Q931: Applying typeplan for sw-type
> 0x12 is 0x2 0x1, Calling num 30210035294
> *Aug 30 09:50:55.631: ISDN Se0/0/0:15 Q931: Sending SETUP  callref =
> 0x0087 callID = 0x8008 switch = primary-net5 interface = User
> *Aug 30 09:50:55.631: ISDN Se0/0/0:15 Q931: TX -> SETUP pd = 8  callref =
> 0x0087
> Bearer Capability i = 0x9090A3
> Standard = CCITT
> Transfer Capability = 3.1kHz Audio
> Transfer Mode = Circuit
> Transfer Rate = 64 kbit/s
> Channel ID i = 0xA9839F
> Exclusive, Channel 31
> Progress Ind i = 0x8181 - Call not end-to-end ISDN, may have
> in-band info
> Calling Party Number i = 0x2181, '30210035294'
> Plan:ISDN, Type:National
> Called Party Number i = 0x81, '0302'
> Plan:ISDN, Type:Unknown
> *Aug 30 09:50:55.631: //84/45469900/SIP/Msg/ccsipDisplayMsg:
> Sent:
> SIP/2.0 100 Trying
> Via: SIP/2.0/TCP 10.0.242.58:5060;branch=z9hG4bK384343afdac
> From:  >;tag=15397~2193bb14-5a60-4057-a1e8-d01a7c16325d-49442666
> To: 
> Date: Thu, 30 Aug 2012 09:50:55 GMT
> Call-ID: 45469900-3f137a3-1c0-3af2000a@10.0.242.58
> CSeq: 101 INVITE
> Allow-Events: telephone-event
> Server: Cisco-SIPGateway/IOS-15.2.4.M1
> Content-Length: 0
>
>
> *Aug 30 09:50:56.495: ISDN Se0/0/0:15 Q931: RX <- SETUP_ACK pd = 8
>  callref = 0x8087
> Channel ID i = 0xA9839F
> Exclusive, Channel 31
> *Aug 30 09:50:57.135: ISDN Se0/0/0:15 Q931: RX <- DISCONNECT pd = 8
>  callref = 0x8087
> Cause i = 0x8283 - No route to destination
> Facility i = 0x91A10802011F0201228100
> Protocol Profile = Remote Operations Protocol
> 0xA10802011F0201228100
> Component = Invoke component
> Invoke Id = 31
> Operation = AOCDChargingUnit
> Progress Ind i = 0x8288 - In-band info or appropriate now
> available
> *Aug 30 09:50:57.135: ISDN Se0/0/0:15 Q931: call_disc: PI received in
> disconnect; Postpone sending RELEASE for callid 0x8008
> *Aug 30 09:50:57.139: //84/45469900/SIP/Msg/ccsipDisplayMsg:
> Sent:
> SIP/2.0 183 Session Progress
> Via: SIP/2.0/TCP 10.0.242.58:5060;branch=z9hG4bK384343afdac
> From:  >;tag=15397~2193bb14-5a60-4057-a1e8-d01a7c16325d-49442666
> To: ;tag=CF900-858
> Date: Thu, 30 Aug 2012 09:50:55 GMT
> Call-ID: 45469900-3f137a3-1c0-3af2000a@10.0.242.58
> CSeq: 101 INVITE
> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
> NOTIFY, INFO, REGISTER
> Allow-Events: telephone-event
> Remote-Party-ID:  >;party=called;screen=no;privacy=off
> Contact: 
> Supported: sdp-anat
> Reason: Q.850;cause=3
> Server: Cisco-SIPGateway/IOS-15.2.4.M1
> Content-Type: application/sdp
> Content-Disposition: session;handling=required
> Content-Length: 256
>
> v=0
> o=CiscoSystemsSIP-GW-UserAgent 7858 2021 IN IP4 10.0.242.61
> s=SIP Call
> c=IN IP4 10.0.242.61
> t=0 0
> m=audio 16398 RTP/AVP 0 8 101
> c=IN IP4 10.0.242.61
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
>
> *Aug 30 09:51:37.203: ISDN Se0/0/0:15 Q931: RX <- RELEASE pd = 8  callref
> = 0x8087
> Cause i = 0x8283 - No route to destination
> Facility i = 0x91A10802011F0201228100
> Protocol Profile = Remote Operations Protocol
> 0xA10802011F0201228100
> Component = Invoke component
> Invoke Id = 31
> Operation = AOCDChargingUnit
> *Aug 30 09:51:37.203: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8
>  callref = 0x0087
> *Aug 30 09:51:37.227:

[OSL | CCIE_Voice] sip gateway PSTN disconnect cause delayed send to cisco phone

[Steffen Bruening]

Hi all,

this question is not ccie lab related. I have 2911 integrated via sip to
cucm 8.6. When a call terminates because of user busy/unallocated number or
what ever, the cisco phone still rings for 30 seconds because the sip
disconnect message was send 30 seconds after pstn disconnect. How could I
reduce this timer?

*Aug 30 09:50:55.631: ISDN Se0/0/0:15 Q931: Applying typeplan for sw-type
0x12 is 0x2 0x1, Calling num 30210035294
*Aug 30 09:50:55.631: ISDN Se0/0/0:15 Q931: Sending SETUP  callref = 0x0087
callID = 0x8008 switch = primary-net5 interface = User
*Aug 30 09:50:55.631: ISDN Se0/0/0:15 Q931: TX -> SETUP pd = 8  callref =
0x0087
Bearer Capability i = 0x9090A3
Standard = CCITT
Transfer Capability = 3.1kHz Audio
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA9839F
Exclusive, Channel 31
Progress Ind i = 0x8181 - Call not end-to-end ISDN, may have
in-band info
Calling Party Number i = 0x2181, '30210035294'
Plan:ISDN, Type:National
Called Party Number i = 0x81, '0302'
Plan:ISDN, Type:Unknown
*Aug 30 09:50:55.631: //84/45469900/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.0.242.58:5060;branch=z9hG4bK384343afdac
From: ;tag=15397~2193bb14-5a60-4057-a1e8-d01a7c16325d-49442666
To: 
Date: Thu, 30 Aug 2012 09:50:55 GMT
Call-ID: 45469900-3f137a3-1c0-3af2000a@10.0.242.58
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M1
Content-Length: 0


*Aug 30 09:50:56.495: ISDN Se0/0/0:15 Q931: RX <- SETUP_ACK pd = 8  callref
= 0x8087
Channel ID i = 0xA9839F
Exclusive, Channel 31
*Aug 30 09:50:57.135: ISDN Se0/0/0:15 Q931: RX <- DISCONNECT pd = 8
 callref = 0x8087
Cause i = 0x8283 - No route to destination
Facility i = 0x91A10802011F0201228100
Protocol Profile = Remote Operations Protocol
0xA10802011F0201228100
Component = Invoke component
Invoke Id = 31
Operation = AOCDChargingUnit
Progress Ind i = 0x8288 - In-band info or appropriate now available
*Aug 30 09:50:57.135: ISDN Se0/0/0:15 Q931: call_disc: PI received in
disconnect; Postpone sending RELEASE for callid 0x8008
*Aug 30 09:50:57.139: //84/45469900/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP 10.0.242.58:5060;branch=z9hG4bK384343afdac
From: ;tag=15397~2193bb14-5a60-4057-a1e8-d01a7c16325d-49442666
To: ;tag=CF900-858
Date: Thu, 30 Aug 2012 09:50:55 GMT
Call-ID: 45469900-3f137a3-1c0-3af2000a@10.0.242.58
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: ;party=called;screen=no;privacy=off
Contact: 
Supported: sdp-anat
Reason: Q.850;cause=3
Server: Cisco-SIPGateway/IOS-15.2.4.M1
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 256

v=0
o=CiscoSystemsSIP-GW-UserAgent 7858 2021 IN IP4 10.0.242.61
s=SIP Call
c=IN IP4 10.0.242.61
t=0 0
m=audio 16398 RTP/AVP 0 8 101
c=IN IP4 10.0.242.61
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

*Aug 30 09:51:37.203: ISDN Se0/0/0:15 Q931: RX <- RELEASE pd = 8  callref =
0x8087
Cause i = 0x8283 - No route to destination
Facility i = 0x91A10802011F0201228100
Protocol Profile = Remote Operations Protocol
0xA10802011F0201228100
Component = Invoke component
Invoke Id = 31
Operation = AOCDChargingUnit
*Aug 30 09:51:37.203: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8
 callref = 0x0087
*Aug 30 09:51:37.227: //84/45469900/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 10.0.242.58:5060;branch=z9hG4bK384343afdac
From: ;tag=15397~2193bb14-5a60-4057-a1e8-d01a7c16325d-49442666
To: ;tag=CF900-858
Date: Thu, 30 Aug 2012 09:50:55 GMT
Call-ID: 45469900-3f137a3-1c0-3af2000a@10.0.242.58
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M1
Reason: Q.850;cause=3
Content-Length: 0
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Re: [OSL | CCIE_Voice] five lab workbook CUCM-DHCP issue

[Steffen Bruening]

Hi,

I don't know whether you are using your own equipment or remote
proctorlabs. You should check your "show ip route" output. Also you can try
to restart the dhcp monitor service or to disable the cisco security agent
with "utils csa disable" (server reboot required).

2012/8/29 Krishna 

> hi folks...
>
> So far i have done 6 labs practicing ipexpert five lab workbook, and
> everytime i encountered the CUCM dhcp issue where site A and Site B phone
> are unable to register due to dhcp issue... on site A router A, and Site B
> router interface's had ip helper-address 10.10.210.10.. when i debup ip
> packet for acl that includes udp ports 67 68.. here is the message i found
> and tried google it but no luck in finding the solution... here is the
> debug that looks like..
>
> FIBipv4-packet-proc: packet routing failed
>
> Note: the dhcp configured  local to router's work fine with no issues...
>
> thank you
> krishna.
>
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>
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Re: [OSL | CCIE_Voice] New Lab 1 CCX no value for calls in queue

[Steffen Bruening]

Hi Edgar,

I know which sites are showing the solution and I have done it that way. I
can see the word "callinq" in my Cdata screen, but there is no colon and no
number (0 or 1).

2012/8/25 Edgar Feliz 

> I think you have to add that look at pages 176-178 on the guide book for
> lab 1.
>
> Edgar
>
> On Sat, Aug 25, 2012 at 5:23 AM, Steffen Bruening wrote:
>
>> Hi all,
>>
>> my ccx scripting skills are only on a basic level. When I configure the
>> script and the layout I never seen a value behind the callinq variable in
>> cdata during an incoming or active call. I test with my own script also the
>> with the script from the solution guide, in both case the behaviour is
>> equal. What I miss or where is the failure in the solution guide?
>>
>> Regards
>>
>> Steffen
>>
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[OSL | CCIE_Voice] New Lab 1 CCX no value for calls in queue

[Steffen Bruening]

Hi all,

my ccx scripting skills are only on a basic level. When I configure the
script and the layout I never seen a value behind the callinq variable in
cdata during an incoming or active call. I test with my own script also the
with the script from the solution guide, in both case the behaviour is
equal. What I miss or where is the failure in the solution guide?

Regards

Steffen
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Re: [OSL | CCIE_Voice] Trunk Configuration - “Media Termination Point Required” Option

[Steffen Bruening]

Hi Raf,

in case you have Gatekeeper involved with a Cube (invia) and you are not
allowed to uncheck the h245 capability flag you could use an MTP for
outbound fast start calls and so yo avoid a capability handshake between
cube and gatekeeper.

2012/8/10 Rafael Chavantes 

> Hello All,
> I'm having a hard time to fully understand the option “Media Termination
> Point Required” under a trunk configuration?
> If i'm not wrong if we have this all call flow will be using the MTP, in
> this case the CUCM, right?
>
> And in the case of a trunk to a Gatekeeper? Since there is no RTP passing
> through the Gatekeeper this option will affect in something? If yes how the
> flow will behave?
>
> Thanks,
> --
> Rafael Chavantes
>
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[OSL | CCIE_Voice] call routing - There isn't just one way of doing it

[Steffen Bruening]

Hi all,

when ever I go through the call routing section I find several ways to
answer the question, some straight, some inconvient. But I don't know for
what I can get the points. Does the way matter to reach the goal/answer the
question (as long as it not vialote the question itself)? Or is irrelevant
as long as the phones ringing as they should and the isdn debug  shows
correct plan and type, channel and calling name?

Regards
Steffen
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Re: [OSL | CCIE_Voice] iDivert/DND in SRST

[Steffen Bruening]

Hi Ramy,

does the question say that you should preserve this?  There is no iDivert
softey in CME/SRST 7.X. iDivert is supported for SIP Phones in CME with 8.5
or so but not for phones falling back to SRST. You can only preserve the
feature with workarounds like this:

Transfer a call directly into cue mailbox
http://www.cisco.com/en/US/products/sw/voicesw/ps5520/products_tech_note09186a00802ab979.shtml


Regards
Steffen

2012/8/7 Ramy Abdelrahim 

>  Dear All,
>
> When the phone is registered to UCM it has iDivert softkey button to
> transfer a call to VM while ringing. When this site goes into SRST, iDivert
> is not there. Do I have to preserve this feature in SRST? And if it's the
> case then how?
>
> Can anyone help on this?
>
> Regards,
> Ramy
>
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[OSL | CCIE_Voice] new lab 3, 5.1 media resources

[Steffen Bruening]

Hi all,

in section 5.1 is written "Configure a transcoder on Site A gateway...Site
A devices *must have the privileges* to use the transcoder when necessary."

Maybe somebody can help me to get a better understanding on this. Does this
mean the Site A is the only site which should allow to use it, or could
also other site use it? How would you interpret this request?

Regards
Steffen
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Re: [OSL | CCIE_Voice] Where all the phones gone?

[Steffen Bruening]

Yes, the first three commands I put in on every device are, cdp run, cdp ad
and cdp t 5.

2012/7/16 Ronmac 

> Hi,
>
> Maybe stupid remark but,
> Is cdp enabled. AdverV2!
> V1 os not ok
>
> Regards ron
>
> Send from my mobile
>
> Op 16 jul. 2012 om 12:41 heeft Steffen Bruening 
> het volgende geschreven:
>
> Hi James,
>
> the ports where the phones should be connected are up and I assume the
> won't be up when no phone/device is attached to them. CDP is enabled but it
> doesn't show cdp information about the devices which are connected to these
> ports. Power Inline Auto is the default power mode and have never seen any
> explicit power command on the 3750 or switch-modules in the routers which
> could affect the power status of these ports.
>
> Regards
>
> Steffen
>
> 2012/7/13 James Dull 
>
>> Steffen,
>>
>> Specifically for BR2-RTR and HQ-3750 the ports may need to have power
>> inline auto applied, shut for about 30 seconds to 1 minute and no shut.
>> BR1-RTR should never need this because it is using a power daughter card
>> which always has power applied to the ports. For the 7962's they are
>> located remotely and I do not directly control which pods they are assigned
>> to but assume that there are not enough to have on every pod at the same
>> time so they can be moved by the voice instructor according to where he
>> needs them. There should always be phones on the devices though as the
>> 7960s are in house and we do not remove those. I ask that you try what I
>> recommended to get the phones to show up, they should immediately show up
>> in show cdp neigh once power inline auto has been applied.
>>
>>
>> James Dull - CCNA, Comp TIA A+, Network+, Security+
>> Technical Support/Technical Editor - IPexpert, Inc., Masonic (MIT), Inc.
>>
>> URL:http://www.IPexpert.com/Tell-Me-More<http://www.ipexpert.com/Tell-Me-More>
>>
>> Phone: +1.810.326.1444 ext 206
>> Email: jd...@ipexpert.com
>> Twitter: www.Twitter.com/IPexpert
>> Check out OUR CATALOG: http://www.ipexpert.com/catalog
>> 
>>
>> On Jul 13, 2012, at 1:47 AM, Steffen Bruening 
>> wrote:
>>
>> Hi all,
>>
>> during all my last 4 proctorlab sessions I missed phones connected to the
>> switch/switch-modules. Often only one phone per site is available,
>> sometimes a complete site has none. It is really really annoying to find a
>> solution about how to bring enough phones (own physical devices or soft
>> clients) to the pod. How are your experiences with that? I am using always
>> the sessions at 12:00 AM PST.
>>
>> Regards
>>
>> Steffen
>> ___
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>> visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com
>>
>>
>>
> ___
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> visit www.ipexpert.com
>
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> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] Where all the phones gone?

[Steffen Bruening]

Hi James,

the ports where the phones should be connected are up and I assume the
won't be up when no phone/device is attached to them. CDP is enabled but it
doesn't show cdp information about the devices which are connected to these
ports. Power Inline Auto is the default power mode and have never seen any
explicit power command on the 3750 or switch-modules in the routers which
could affect the power status of these ports.

Regards

Steffen

2012/7/13 James Dull 

> Steffen,
>
> Specifically for BR2-RTR and HQ-3750 the ports may need to have power
> inline auto applied, shut for about 30 seconds to 1 minute and no shut.
> BR1-RTR should never need this because it is using a power daughter card
> which always has power applied to the ports. For the 7962's they are
> located remotely and I do not directly control which pods they are assigned
> to but assume that there are not enough to have on every pod at the same
> time so they can be moved by the voice instructor according to where he
> needs them. There should always be phones on the devices though as the
> 7960s are in house and we do not remove those. I ask that you try what I
> recommended to get the phones to show up, they should immediately show up
> in show cdp neigh once power inline auto has been applied.
>
>
> James Dull - CCNA, Comp TIA A+, Network+, Security+
> Technical Support/Technical Editor - IPexpert, Inc., Masonic (MIT), Inc.
>
> URL:http://www.IPexpert.com/Tell-Me-More<http://www.ipexpert.com/Tell-Me-More>
>
> Phone: +1.810.326.1444 ext 206
> Email: jd...@ipexpert.com
> Twitter: www.Twitter.com/IPexpert
> Check out OUR CATALOG: http://www.ipexpert.com/catalog
>
> On Jul 13, 2012, at 1:47 AM, Steffen Bruening 
> wrote:
>
> Hi all,
>
> during all my last 4 proctorlab sessions I missed phones connected to the
> switch/switch-modules. Often only one phone per site is available,
> sometimes a complete site has none. It is really really annoying to find a
> solution about how to bring enough phones (own physical devices or soft
> clients) to the pod. How are your experiences with that? I am using always
> the sessions at 12:00 AM PST.
>
> Regards
>
> Steffen
> ___
> 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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[OSL | CCIE_Voice] Where all the phones gone?

[Steffen Bruening]

Hi all,

during all my last 4 proctorlab sessions I missed phones connected to the
switch/switch-modules. Often only one phone per site is available,
sometimes a complete site has none. It is really really annoying to find a
solution about how to bring enough phones (own physical devices or soft
clients) to the pod. How are your experiences with that? I am using always
the sessions at 12:00 AM PST.

Regards

Steffen
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[OSL | CCIE_Voice] New Lab 2, 8.2 tracing UCM<>CUPS signaling, missing picture in DSG

[Steffen Bruening]

Hi all,

there is a picture missing on page 197 in the solution guide, which should
show how to setup the trace settings. Can somebody explain me how I should
setup the trace?

Regards

Steffen
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[OSL | CCIE_Voice] Ipexpert CCIE Voice Five Lab Handbook

[Steffen Bruening]

Hi all,

I have the BLS from IPexpert and I feel very comfortable with the 10 labs
of it. Now I got a marketing email for 5 extra labs made by Ipexpert to
learn for the real lab.

www.ipexpert.com/cisco/ccie/voice/handbook

Does somebody of you buyed this product? Is it it worth? What are the
differents compared to the 10 "old" labs? Are they harder/more difficult?

Regards

Steffen
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Re: [OSL | CCIE_Voice] Calling Number during Unity Connection Call Transfer to external number

[Steffen Bruening]

 

I found it,

 

CUCM Service Parameter:

 

Display Original Calling Number on Transfer from Cisco Unity – Value needs to 
be set to true.

 

Von: Cristobal Priego [mailto:cristobalpri...@gmail.com] 
Gesendet: Donnerstag, 15. September 2011 03:55
An: Steffen Bruening
Cc: 
Betreff: Re: [OSL | CCIE_Voice] Calling Number during Unity Connection Call 
Transfer to external number

 

I there is a service parameter that you have to chante to preserve the caller 
id. I don't remember the name of the option though

Hth

Enviado desde mi iPhone


El Sep 14, 2011, a las 14:21, "Steffen Bruening" 
 escribió:

Hi all,

 

this question is not related to the CCIE labs, but hopefully you can 
help me to find an answer.

 

When a caller reach a user voicemail and choose the caller input option 
to be forwarded to an alternate number (in this case the mobile number) I can 
see in the isdn debug that extension of the used voicemail-port for this call 
transfer is used as calling number instead of the originating number. Any ideas 
how I could achieve that the originating phone number will be presented? The 
german telco do not check or restrict the calling number presentation (CLIP no 
screening).

 

Regards

Steffen

 


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[OSL | CCIE_Voice] Calling Number during Unity Connection Call Transfer to external number

[Steffen Bruening]

Hi all,

 

this question is not related to the CCIE labs, but hopefully you can
help me to find an answer.

 

When a caller reach a user voicemail and choose the caller input option
to be forwarded to an alternate number (in this case the mobile number)
I can see in the isdn debug that extension of the used voicemail-port
for this call transfer is used as calling number instead of the
originating number. Any ideas how I could achieve that the originating
phone number will be presented? The german telco do not check or
restrict the calling number presentation (CLIP no screening).

 

Regards

Steffen

 


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[OSL | CCIE_Voice] mixture of own and remote hardware for vRack session

[Steffen Bruening]

Hi all,

 

I have made two vRack sessions so far based on Remote Phones and
Softphones. For my next session I want to use the hardware vpn session.
I have a 2811 with 4-Port Hwic and 3 7962 phones, but as written on
proctorlabs it seems that this is not enough (requirements are 5 phones
and a 8-port switch) . Isn`t it possible/allowed to use a mixture of my
own phones and some of the remote phones?

 

Regards

Steffen


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