[OSL | CCIE_Voice] Provide outside dial-tone confusion !

[Cisco Voip]

Hi all

I am using CUCM 7 and in my lab environment, i have configured the following 
route patterns.

600.00!         
600.03X                
600.0600X            
600.0[24-9]!            
600.111XX                
600.81X        
600.[39]XXX    

( All patterns are giving valid gateway address, provide outside dial tone is 
checked and urgent prority is set. No translation pattern is being used)

Following is the sequence of steps

1) I dial 6 (no outside tone)
2) then i dial 0 (i.e. 60) now i hear outside tone

What is the logic ? i mean why i didnt hear secondary dial tone on 6 ??
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[OSL | CCIE_Voice] FXO lines !

[Cisco Voip]

Hi all. 

I am using cisco 2801 with 2 fxo cards. Running CME 
version 7. I am using 7940 phone. Now is there anyway that when the user
 of 7940 press line 1, he gets FXO line 1, and when he press 2 he gets 
FXO line 2.

Is this possible or do i need to define outside access code for both these 
lines ?
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[OSL | CCIE_Voice] SIP trunk help !

[Cisco Voip]

Hi all. 

I am setting up a sip trunk with a local ITSP. Now they  havent provided me 
with 
any username and password. Can someone tell me  what configuration do i need to 
do on router ?

My router is 2821 with 12.4(22)T adventerprise


They have just provided me the sip server address and range of DID with first 
number being our primary number.___
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[OSL | CCIE_Voice] E1 Trunk

[Cisco Voip]

Hi all.
 
lets say i have 1 head office and 3 branches. All 3 branches are using CME. 
Headoffice has E1 from local telco. Now the requirement is,
 
When any person in any of 3 branches, dials 9 (access code), one of the E1 
channels should be allocated to him !!! (if none are free, then he should hear 
busy tone)
 
Can someone give me some idea as to whether this is even possible ?
 
Note: (HQ and all branches are running CME 7.x)


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[OSL | CCIE_Voice] CUCM 8 call blocking

[Cisco Voip]

Hi all.

Is there a way of blocking inbound/outbound calls on CUCM 8/7 ? during my 
research i found conflicting facts like, it can only be done on gateway running 
H323 only ! so i am a bit confused. 


Kindly guide me


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[OSL | CCIE_Voice] CME concept of partition and search spaces !

[Cisco Voip]

Hi all.

In my office, i have 10 executives, whom we want only selected group of people 
to call. By default, anyone can call them right ? In cucm we are able to do 
this 
via CSS. How can we achieve the same in CUCME ?



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[OSL | CCIE_Voice] OT: IPexpert CCNP voice !

[Cisco Voip]

Hi all. 

I am new here and i will certainly make sure i wont right any OT from here on 
so 
consider this my first and last one. Since IPexpert instructors are also here, 
do you guys are planning to launch CCNP voice vods and workbooks ? 


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Re: [OSL | CCIE_Voice] Forward only 4 digits

[Cisco Voip]

Thanks alot Sir :-)





From: Shrini 
To: Cisco Voip 
Cc: ccie_voice@onlinestudylist.com
Sent: Sat, January 29, 2011 10:59:05 PM
Subject: RE: [OSL | CCIE_Voice] Forward only 4 digits

 
For  POTS dial-peer  it strips all digits explicitly defined. Since you are  
saying it is sending all digits I assume it is a VOIP  dialpeer.
In  this case you have to use translation rule.
 

On Sat, Jan 29, 2011 at 1:19 PM, Cisco Voip  wrote:

Hi all 
>
>
>My setup is simple
> 
>(ext 1001)
>IPc1R1R2-IPc2
>(ext 2001)
> 
>
>I have activated secondary dial tone feature and the digit is 9.
> 
>Now the destinaton-pattern on R1 is
> 
>destinaton-pattern 920..
>
>But now, the call doesnt connect, when i ran the debug, it showed methat 
>R1 
>is forwarding 92001 instead of 2001, which is logical. How can i forceR1 
>to 
>strip 9 and forward only 2001 ?
> 
>Do i have to use translation rules here ? is there any other method?
>
>
>___
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>visit www.ipexpert.com
>
>



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[OSL | CCIE_Voice] Forward only 4 digits

[Cisco Voip]

Hi all 


My setup is simple
 
(ext 1001) 
IPc1R1R2-IPc2
 (ext 2001)
 

I have activated secondary dial tone feature and the digit is 9.
 
Now the destinaton-pattern on R1 is
 
destinaton-pattern 920..

But now, the call doesnt connect, when i ran the debug, it showed me that R1 is 
forwarding 92001 instead of 2001, which is logical. How can i force R1 to strip 
9 and forward only 2001 ?
 
Do i have to use translation rules here ? is there any other method ?



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Re: [OSL | CCIE_Voice] Call Forward Unregistered

[cisco voip]

Hi Mark Holloway,

What bug are you talking about. Do you have a bug id?

On Mon, Oct 18, 2010 at 3:36 PM, sisiaji  wrote:

> hey guys, i truly have no clue what you are talking about :))) what VM has
> to do with CFUR?
>
> For and By fields are representing what you have configured as redirecting
> Calling Party number mask (in this case redirecting ip phone) and what you
> have configured as a destination for such calls (Unregistered). if both are
> set with +... then both will be shown as +... in For/By fields... it is not
> a rocket science I would say so...
>
> however, for CFUR, you have to be extremely careful, as it doesn't require
> separate partitions/CSS to work, but if you think about it, it is the only
> way to fine tune it to what you want.
>
> so don't overcomplicate it, set your Unregistered destination to be
> +19723033001 and assuming your calling party mask is already the same, then
> you just need to create RP for the same + number inside separate partition
> which will be the only one present in a separate CSS, which in turn will
> need to be assigned to Unregistered Destination CSS. nothing else.
> when you create RP for +..., you just need to do proper digit manipulation
> depending on which location gateway calls is supposed to go out. so if this
> is national call, then you have to put inside RG/RL manipulation pre-dot
> (for +1.972XX), called type National, plan isdn) and don't touch calling
> party xformations at all as by default they are set on callmanager which
> means only internal 4 digits will be sent as calling numbers (that is what
> you see inside brackets).
>
> ok? :)
>
>
>
> On Mon, Oct 18, 2010 at 1:51 AM, Mark Holloway wrote:
>
>> I think the main thing to understand is that it should work using E164 in
>> For/By under normal circumstances and everything else we are suggesting is a
>> work around to a known bug with CUCM 7.0 and VMWare.
>>
>>
>> On Oct 17, 2010, at 3:56 PM, Daniel Berlinski wrote:
>>
>> Hello guys
>>
>> If you want to manipulate this with CUCM the place to change the
>> redirected number is the VM profile as indicated by Mark.  Alternatively you
>> could attach an additional rule to the translation-profile plugged inbound
>> to the POTS call leg in the branch router in SRST mode and configure it to
>> change the redirect-called number from  to the e164 that you are after.
>>
>> Cheers
>>
>> On Mon, Oct 18, 2010 at 11:36 AM, Mark Holloway wrote:
>>
>>> I ran into this same issue. Supposedly it's a problem with CUCM 7.0 and
>>> VMWare.  If you go to the Device > Phone and click on the Site B phones >
>>> Line and specifically assign the Voicemail Profile to the Line it might
>>> work.  I had success a couple of times doing this, but then after resetting
>>> my rack the last time and assigning the VM profile to the Line I still had
>>> this issue.
>>>
>>> On Oct 17, 2010, at 3:28 PM, Afzal Bhutta wrote:
>>>
>>> Scenario:
>>>
>>> In SRST mode: HQ MGCP gateway and Site-B H322 gateway Site-C H322 gateway
>>> cme
>>>
>>> HQ and Site C phones are being able to call SiteB Phone 1 using 4 digits
>>> dialing in SRST.(Wan failure)
>>>
>>> I use call forward unregistered feature.
>>> When I call from HQ Phone-1 call routed through HQ Gateway.
>>> When I call from Site-C Phone-1 call routed through the GK first and then
>>> HQ Gateway.
>>> Below is the display I am getting on my Site-B phone display.
>>>
>>>
>>> Forward HQ Phone 1
>>> (2001)
>>> For   3001
>>> By3001
>>>
>>>
>>> Forward Site-C Phone 1
>>> (4001)
>>> For   3001
>>> By3001
>>>
>>>
>>> My question how can I achieve below display in FOR and BY field it should
>>> be E.164 number format and than 4 digits internal ID
>>>
>>>
>>>
>>>
>>> Forward
>>> (2001)
>>> For   +19723033001 (3...)
>>> By+19723033001 (3...)
>>> Forward
>>> (4001)
>>> For   +19723033001 (3...)
>>> By+19723033001 (3...)
>>>
>>>
>>> Thanking you in anticipation folks.
>>> ___
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>>> visit www.ipexpert.com
>>>
>>>
>>>
>>> ___
>>> For more information regarding industry leading CCIE Lab training, please
>>> visit www.ipexpert.com
>>>
>>>
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
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Re: [OSL | CCIE_Voice] Hi Experts (DHCP lease)

[cisco voip]

if your dhcp lease is 24 hours, it does not mean that lease will expire
exactly 24 hours later and if dhcp server is not reachable then IP Address
will be released.

if DHCP lease is 24 hours, your phone will query the DHCP server after 12
hours to see if server is reachable, if it does the lease will refresh for
another 24 hours. If it is not available phone will query again after 6
hours then 3 hours then 1.5 hours and so on.

my recommendation is do not change anything if it is not necessary.

On Mon, Oct 18, 2010 at 7:10 PM, Pithog Oil  wrote:

> In the actual lab is it a good idea to Set my DHCP lease times to several
> days, or use a local DHCP server to ensure that IP Phones do not reset
> themselves when their lease expires during a WAN outage.
>
> Can i be penalised for doing this.
>
>
> ___
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> visit www.ipexpert.com
>
>
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[OSL | CCIE_Voice] Buffers, Queue

[cisco voip]

Hi,

Can someone please explain me the difference between buffers and queue and
how they are implemented on 3750.
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Re: [OSL | CCIE_Voice] CHALLENGES WITH CUE ! ! !

[cisco voip]

"WAIT WHILE I TRANSFER YOUR CALL" is transfer greeting and you don't
transfer a call to send the message. Can you please explain how are you
sending the message

On Mon, Sep 13, 2010 at 10:18 AM, Pithog Oil  wrote:

> Hi Experts
>
> I configured CUE and it works properly, except for this behaviour which i
> dont think is the normal behaviour, i get this after sending a voice mail,
> WAIT WHILE I TRANSFER YOUR CALL then suddenly the phone that timed out
> without picking the call starts ringing. whearas my Unity connection does
> not have this behaviour.
>
> Also for VPIM my message was sent on CUE agent only  to see them return
> back to my phone after a few minutes, However from Cisco Unity i could not
> get messages sent accros, i am never prompted to enter my location ID, i
> know i am missing a critical bit, *i need an expert to lighten me up on
> how to troubleshoot VPIM effectively.*
> **
> Thanks in anticipation
>
>
> ___
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> visit www.ipexpert.com
>
>
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Re: [OSL | CCIE_Voice] VM config for SIP phones in CME/CUE

[cisco voip]

Hey Roger,

Configuring voice class codec under voice register pool is the only way i
could think of to make SIP phone to work with both CUE and to talk over wan.


If you don't configure codec under voice register pool, it will take g729
and will not work with CUE, if you configure codec g711ulaw, call will not
work over wan.

and voice class codec always worked for me for SIP CME.

Comments please.


2010/9/11 Roger Källberg 

>  Although the command voice class codec is possible to use on a voice
> register pool it will not take effect. That's one of the small odditiesthat
> you simply need to be aware of when dealing with SIP CME phones.
>
> Sincerely
>
>  *Roger Källberg*
> CCIE #26199 (Voice)
> Consultant
> Cygate AB
> Eric Perssons väg 21, SE-217 62 MALMÖ
>   --
> *Från:* cisco voip [voip.ccieci...@gmail.com]
> *Skickat:* den 11 september 2010 09:01
> *Till:* Randall Saborio
> *Kopia:* ccie_voice@onlinestudylist.com; Avinash Shukla
> *Ämne:* Re: [OSL | CCIE_Voice] VM config for SIP phones in CME/CUE
>
>  Also voice register pool should have a voice class codec defined, which
> has both g711 and g729 in it
>
> On Sat, Sep 11, 2010 at 3:03 AM, Randall Saborio wrote:
>
>> Another imoprtant one:
>>
>> voice service voip
>>   allow-connections sip to sip
>>
>>
>>
>>
>> On Fri, Sep 10, 2010 at 7:22 AM, Amy Ryan  wrote:
>>
>>> Avinash,
>>>
>>> Here is an example configuration for a SIP phone covering to VM in CUCME.
>>>
>>> *
>>> sip-ua
>>>  mwi-server ipv4:10.10.202.2>> server>
>>> !
>>> voice register dn  2
>>>  call-forward b2bua busy 3600
>>>  call-forward b2bua noan 3600 timeout 12>> VM pilot>
>>> mwi 
>>> !
>>> voice register pool  2
>>>  dtmf-relay rtp-nte
>>> !
>>> voice register global
>>>  voicemail 3600 >> the phone>
>>> create profile
>>>  reset
>>> !
>>> dial-peer voice 3600 voip
>>>  destination-pattern 3600
>>>  session protocol sipv2
>>>  session target ipv4:10.10.202.2
>>>  dtmf-relay sip-notify
>>>  codec g711ulaw
>>>  no vad
>>>  incoming called-number 399[89]….
>>> *
>>> HTH,
>>> Amy
>>>
>>>
>>>
>>> ---
>>> Amy Ryan – CCIE #24677 (Voice)
>>> Technical Instructor - IPexpert, Inc.
>>> Mailto: *ar...@ipexpert.com
>>> *Telephone: +1.810.326.1444
>>> Live Assistance, Please visit: www.ipexpert.com/chat <*
>>> http://www.ipexpert.com/chat*>
>>> eFax: +1.810.454.0130
>>>
>>> IPexpert is a premier provider of Self-Study Workbooks, Video on Demand,
>>> Audio Tools, Online Hardware Rental and Classroom Training for the Cisco
>>> CCIE (R&S, Voice, Wireless, Security & Service Provider) certification(s)
>>> with training locations throughout the United States, Europe, South Asia and
>>> Australia. Be sure to visit our online communities at
>>> www.ipexpert.com/communities <*http://www.ipexpert.com/communities*>
>>>  and our public website at www.ipexpert.com <*http://www.ipexpert.com/*>
>>>
>>>
>>>
>>>
>>> --
>>> *From: *Avinash Shukla 
>>> *Date: *Fri, 10 Sep 2010 18:20:56 +0530
>>> *To: *
>>> *Subject: *[OSL | CCIE_Voice] VM config for SIP phones in CME/CUE
>>>
>>>
>>> Hi Experts,
>>>
>>> I wanted to know what configs do we need to do on the cme/cue to make VM
>>> work for SIP phones.
>>>
>>> I have added user to CUE with extension and have also mentioned the
>>> Voicemail DN command in voice register pool. + the dial peer for voicemail.
>>> When i do debug voip dialpeer all. I can see the dialpeer for Voicemail
>>> getting hit but i get nothin but the call does not completes!
>>>
>>> Voice mail works just fine for SCCP phones but not for SIP.
>>>
>>> Regards,
>>> Avinash
>>>
>>>  --
>>> ___
>>> For more information regarding industry leading CCIE Lab training, please
>>> visit www.ipexpert.com
>>>
>>> ___
>>> For more information regarding industry leading CCIE Lab training, please
>>> visit www.ipexpert.com
>>>
>>>
>>
>>
>>  --
>> Randall "da ill" Saborio
>> CCIE Voice Wannabe #10054675811
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>
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Re: [OSL | CCIE_Voice] VM config for SIP phones in CME/CUE

[cisco voip]

Also voice register pool should have a voice class codec defined, which has
both g711 and g729 in it

On Sat, Sep 11, 2010 at 3:03 AM, Randall Saborio  wrote:

> Another imoprtant one:
>
> voice service voip
>   allow-connections sip to sip
>
>
>
>
> On Fri, Sep 10, 2010 at 7:22 AM, Amy Ryan  wrote:
>
>>  Avinash,
>>
>> Here is an example configuration for a SIP phone covering to VM in CUCME.
>>
>> *
>> sip-ua
>>  mwi-server ipv4:10.10.202.2> server>
>> !
>> voice register dn  2
>>  call-forward b2bua busy 3600
>>  call-forward b2bua noan 3600 timeout 12> pilot>
>>  mwi 
>> !
>> voice register pool  2
>>  dtmf-relay rtp-nte
>> !
>> voice register global
>>  voicemail 3600 > phone>
>>  create profile
>>  reset
>> !
>> dial-peer voice 3600 voip
>>  destination-pattern 3600
>>  session protocol sipv2
>>  session target ipv4:10.10.202.2
>>  dtmf-relay sip-notify
>>  codec g711ulaw
>>  no vad
>>  incoming called-number 399[89]….
>> *
>> HTH,
>> Amy
>>
>>
>>
>> ---
>> Amy Ryan – CCIE #24677 (Voice)
>> Technical Instructor - IPexpert, Inc.
>> Mailto: *ar...@ipexpert.com
>> *Telephone: +1.810.326.1444
>> Live Assistance, Please visit: www.ipexpert.com/chat <*
>> http://www.ipexpert.com/chat*>
>> eFax: +1.810.454.0130
>>
>> IPexpert is a premier provider of Self-Study Workbooks, Video on Demand,
>> Audio Tools, Online Hardware Rental and Classroom Training for the Cisco
>> CCIE (R&S, Voice, Wireless, Security & Service Provider) certification(s)
>> with training locations throughout the United States, Europe, South Asia and
>> Australia. Be sure to visit our online communities at
>> www.ipexpert.com/communities <*http://www.ipexpert.com/communities*>  and
>> our public website at www.ipexpert.com <*http://www.ipexpert.com/*>
>>
>>
>>
>> --
>> *From: *Avinash Shukla 
>> *Date: *Fri, 10 Sep 2010 18:20:56 +0530
>> *To: *
>> *Subject: *[OSL | CCIE_Voice] VM config for SIP phones in CME/CUE
>>
>>
>> Hi Experts,
>>
>> I wanted to know what configs do we need to do on the cme/cue to make VM
>> work for SIP phones.
>>
>> I have added user to CUE with extension and have also mentioned the
>> Voicemail DN command in voice register pool. + the dial peer for voicemail.
>> When i do debug voip dialpeer all. I can see the dialpeer for Voicemail
>> getting hit but i get nothin but the call does not completes!
>>
>> Voice mail works just fine for SCCP phones but not for SIP.
>>
>> Regards,
>> Avinash
>>
>> --
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>
>
> --
> Randall "da ill" Saborio
> CCIE Voice Wannabe #10054675811
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
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Re: [OSL | CCIE_Voice] How device pool affect media resources?

[cisco voip]

Dear David,

He has provided same MRGL in both DP. It means both DP can see Xcoder, but
until both SIP Trunk and Device are in same DP, xcoder is not invoking.
Ki Wi- Can you please try by specifying MRGL on the device that requires it

On Wed, Sep 8, 2010 at 8:25 PM, David Lee  wrote:

> Hi Ki Wi,
>
> The MRGL assigned to a DP is used by the device that is in the DP.
>  Therefore, your observation is accurate.  SIP trunk in HQ DP will use the
> transcoder in the MRGL in the HQ DP.  If the MRGL of the HQ DP does not have
> transcoder, then the SIP trunk does not have access to a transcoder.
>
> As for MoH working, it's probably because your MOH servers are not in any
> MRG, thus they are in the null MRG, which is accessible by anyone looking
> for MOH.
>
> Thanks,
>
> -Dave
>
>
> On Wed, Sep 8, 2010 at 10:49 AM, 
> wrote:
>
>> Send CCIE_Voice mailing list submissions to
>>ccie_voice@onlinestudylist.com
>>
>> To subscribe or unsubscribe via the World Wide Web, visit
>>http://onlinestudylist.com/mailman/listinfo/ccie_voice
>> or, via email, send a message with subject or body 'help' to
>>ccie_voice-requ...@onlinestudylist.com
>>
>> You can reach the person managing the list at
>>ccie_voice-ow...@onlinestudylist.com
>>
>> When replying, please edit your Subject line so it is more specific
>> than "Re: Contents of CCIE_Voice digest..."
>>
>>
>> Today's Topics:
>>
>>   1. How device pool affect media resources? (Ki Wi)
>>   2. Re: Gatekeeper call routing between BR2 CME and BR1 H323
>>  gateway (Tam Nhu)
>>   3. Re: Fast busy when calling from PSTN to BR1 phone..
>>  (chase mergenthal)
>>   4. Re: Fast busy when calling from PSTN to BR1 phone..
>>  (Wilson Bolanos)
>>
>>
>> --
>>
>> Message: 1
>> Date: Wed, 8 Sep 2010 17:18:37 +0800
>> From: Ki Wi 
>> To: ccie_voice@onlinestudylist.com
>> Subject: [OSL | CCIE_Voice] How device pool affect media resources?
>> Message-ID:
>>
>> Content-Type: text/plain; charset="iso-8859-1"
>>
>> I'm currently doing V1 Lab 5C where Q5.2 requires a transcoder.
>>
>> If i placed the transcoder in DP_HQ while i purposely put the SIP trunk
>> into
>> another DP (such as DP_BR1), it fails to be triggered.
>> However, If i placed *both the SIP trunk and the transcoder are in the
>> same
>> DP, it works. *
>> **
>> *  *All my DP contains the same MGRL* *
>>
>> Now, i'm wondering this rule only applies to transcoder or all the other
>> media resources? From what i see, MOH don't seems to have this limitation.
>> -- next part --
>> An HTML attachment was scrubbed...
>> URL:
>> 
>>
>> --
>>
>> Message: 2
>> Date: Wed, 8 Sep 2010 07:20:42 -0500
>> From: Tam Nhu 
>> To: Vik Malhi 
>> Cc: OSL Group 
>> Subject: Re: [OSL | CCIE_Voice] Gatekeeper call routing between BR2
>>CME and BR1 H323 gateway
>> Message-ID:
>>
>> 
>> >
>> Content-Type: text/plain; charset="iso-8859-1"
>>
>> Hi Vik,
>>
>> Thank you for your input.  I saved my configurations for this lab, and
>> have
>> been working on the +dialing lab 10, so let me revert back to this lab
>> tonight and try your suggestions.  I remembered I did unchecked the
>> Outbound
>> Fast Start at one point during troubleshooting, but it did not make any
>> improvements.  I will try again tonight and reply back with results as
>> soon
>> as I can.
>>
>> Thanks,
>> TN.
>> -- next part --
>> An HTML attachment was scrubbed...
>> URL:
>> 
>>
>> --
>>
>> Message: 3
>> Date: Wed, 8 Sep 2010 09:24:27 -0500
>> From: chase mergenthal 
>> To: , ccie voice 
>> Subject: Re: [OSL | CCIE_Voice] Fast busy when calling from PSTN to
>>BR1 phone..
>> Message-ID: 
>> Content-Type: text/plain; charset="windows-1252"
>>
>>
>> It says "TEI_ASSIGNED"; I think i have the config correct in call manager
>> and on the GW.
>>
>> BR1-RTR#sho isdn status
>> Global ISDN Switchtype = primary-ni
>>
>> %Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output may
>> not apply
>>
>> ISDN Serial0/0/0:23 interface
>>dsl 0, interface ISDN Switchtype = primary-ni
>>L2 Protocol = Q.921 0x  L3 Protocol(s) = CCM MANAGER 0x0003
>>Layer 1 Status:
>>ACTIVE
>>Layer 2 Status:
>>TEI = 0, Ces = 1, SAPI = 0, State = TEI_ASSIGNED
>>Layer 3 Status:
>>0 Active Layer 3 Call(s)
>>Active dsl 0 CCBs = 0
>>The Free Channel Mask:  0x8007
>>Number of L2 Discards = 0, L2 Session ID = 2
>>Total Allocated ISDN CCBs = 0
>> BR1-RTR#
>>
>>
>> Parts of config:
>>
>>  interface Serial0/0/0:23
>>  no ip address
>>  encapsulation hdlc
>>  isdn switch-type primary-ni
>>  isdn incoming-voice voice
>>  isdn bind-l3 ccm-manager
>>  no cdp enable
>>
>> controller T1 0/0/0
>>  framing esf
>>  linecode b8zs
>>  pri-group timeslots 1-3,24 service mgcp
>>
>> ccm-manager switchback immediate
>

Re: [OSL | CCIE_Voice] How device pool affect media resources?

[cisco voip]

Can you test by specifying the MRGL on SIP trunk and/or device.

On Wed, Sep 8, 2010 at 2:48 PM, Ki Wi  wrote:

> I'm currently doing V1 Lab 5C where Q5.2 requires a transcoder.
>
> If i placed the transcoder in DP_HQ while i purposely put the SIP trunk
> into another DP (such as DP_BR1), it fails to be triggered.
> However, If i placed *both the SIP trunk and the transcoder are in the
> same DP, it works. *
> **
> *  *All my DP contains the same MGRL* *
>
> Now, i'm wondering this rule only applies to transcoder or all the other
> media resources? From what i see, MOH don't seems to have this limitation.
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] CME Presence Testing and configuration

[cisco voip]

Don't know what you mean by
When Ph-1 line 4001 is on the phone we should see the status of this call in
the local directory of Ph-1.

if someone is on the phone, why it will go into the local directory to see
if he is on the call or not.

The command you are missing is "*ip http server*", w/o this you will not be
able to access local directory on phone



On Sun, Aug 8, 2010 at 11:18 AM, Afzal Bhutta wrote:

> Hi folks,
> My question is how Can I test below feature, (I mean testing and is there
> any command I am missing)
> When Ph-1 line 4001 is on the phone we should see the status of this call
> in the local directory of Ph-1.
> My config is below,
> !
> telephony-service
>  sdspfarm units 2
>  sdspfarm transcode sessions 3
>  sdspfarm tag 1 SCCODER
>  sdspfarm tag 2 SCCONF
>  conference hardware
>  max-ephones 5
>  max-dn 25 no-reg
>  ip source-address 142.102.66.254 port 2000
>  time-zone 42
>  voicemail 4220
>  max-conferences 8 gain -6
>  transfer-system full-consult
>  directory entry 1 4001 name Site C phone 1
>  directory entry 2 4002 name Site C phone 2
>  create cnf-files version-stamp 7960 Mar 03 2002 02:45:49
> !
>
> !
> presence
>  presence call-list
> !
> sip-ua
>  presence enable
> !
> !
> !
> ephone-dn  1  octo-line
>  number 4001 no-reg primary
>  description +85224044001
>  name Site C Phone 1
>  allow watch
>  call-forward busy 4220
>  call-forward noan 4220 timeout 20
>  huntstop channel 1
> !
> !
> ephone-dn  2  octo-line
>  number 4002 no-reg primary
>  description +85224044002
>  name Site C Phone 2
>  call-forward busy 4220
>  call-forward noan 4220 timeout 20
>  huntstop channel 1
> !
> !
> ephone-dn  3
>  number A4001 no-reg primary
>  intercom A4002 label "Intercom-4002"
> !
> !
> ephone-dn  4
>  number A4002 no-reg primary
>  intercom A4001 barge-in label "Intercom-4001"
> !
>
> !
> ephone  1
>  device-security-mode none
>  mac-address 000F.2309.C21F
>  ephone-template 1
>  presence call-list
>  type 7960
>  button  1:1 2:3
> !
> !
> !
> ephone  2
>  device-security-mode none
>  mac-address 0012.01AD.3761
>  ephone-template 1
>  presence call-list
>  blf-speed-dial 1 4001 label "BLF-4001"
>  type 7970
>  button  1:2 2:4
> !
> !
> !
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] Cbarge and SRST (Again) (Again) and (Again) what am I missing???

[cisco voip]

If you press the remote-in use button and you are into conference, it means
you are using single button cbarge.
If you are presented with a remote-in-use softkeys when you press the button
and you have to select cbarge softkey. It means you are using normal cbarge

On Sun, Aug 8, 2010 at 4:23 AM, Mark Holloway  wrote:

> In UCM how do you determine whether you are assigning single button cBarge
> or normal cBarge?
>
>
> On Aug 7, 2010, at 9:35 AM, cisco voip wrote:
>
> That bug is for srst mode auto provision none.. for provision all, it
> should work
>
> The problem you are facing of having cbarge for split second is because you
> had single button cbarge when phones were registered to CUCM, disable that
> setting and make it normal cbarge, they will start to work in srst mode as
> well
>
>
>
> On Fri, Aug 6, 2010 at 5:05 PM, Ashar Siddiqui wrote:
>
>>  I am glad that the solution proposed by Cisco is exactly what I did
>> months back after trying different solutions.
>>
>>
>> Ash.
>>
>>
>> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
>> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *CCIE Voice GMAIL
>> *Sent:* 06 August 2010 03:13
>>
>> *To:* ccie_voice@onlinestudylist.com
>> *Subject:* Re: [OSL | CCIE_Voice] Cbarge and SRST (Again) (Again) and
>> (Again) what am I missing???
>>
>>
>>
>> I thought I’d share this with everyone as this have been extremely
>> frustrating for me.  Apparently this is a known bug (well…recently known).
>>
>>
>> CSCti11843<http://tools.cisco.com/Support/BugToolKit/search/getBugDetails.do?method=fetchBugDetails&bugId=CSCti11843>
>>
>>
>>
>> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
>> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *MARSHALL, JODY C
>> (ATTBCS)
>> *Sent:* Wednesday, August 04, 2010 4:55 AM
>> *To:* ccie_voice@onlinestudylist.com
>> *Subject:* [OSL | CCIE_Voice] Cbarge and SRST (Again) (Again) and (Again)
>> what am I missing???
>>
>>
>> I have read (several) post on this and have tested several different ways.
>> None of which have I been able to make work. Can you please take a look and
>> see if I am missing something. The first configuration is from
>> auto-provision all. I had the phones registers then unregister bounced the
>> router and register again. Cbarge does not work. I see the remote-in-use
>> state for a second then disappears. I then registered the phones to CUCM and
>> removed telephony-service reloaded the router and reconfigured
>> telephony-service with auto-provision none with the second configuration
>> posted. Cbarge does not work.
>>
>> 124-20.T5.bin
>>
>> telephony-service
>>
>>  sdspfarm units 5
>>
>>  sdspfarm tag 2 sitebcfb
>>
>>  conference hardware
>>
>>  srst mode auto-provision all
>>
>>  srst ephone template 1
>>
>>  srst ephone description SRST : Aug 03 2010 13:28:28 : Aug 03 2010
>> 21:20:20
>>
>>  srst dn template 1
>>
>>  srst dn line-mode octo
>>
>>  max-ephones 4
>>
>>  max-dn 30 preference 3
>>
>>  ip source-address 10.12.202.1 port 2000
>>
>>  system message CCIEVOICE
>>
>>  time-zone 8
>>
>>  date-format dd-mm-yy
>>
>>  voicemail 2220
>>
>>  max-conferences 8 gain -6
>>
>>  web admin system name administrator password ccievoice
>>
>>  transfer-system full-consult
>>
>>  transfer-pattern .T
>>
>>  create cnf-files version-stamp 7960 Aug 03 2010 21:20:26
>>
>> !
>>
>>
>> R2#sho sccp
>>
>> SCCP Admin State: UP
>>
>> Gateway IP Address: 10.12.202.1, Port Number: 2000
>>
>> IP Precedence: 5
>>
>> User Masked Codec list: None
>>
>> Call Manager: 10.12.202.1, Port Number: 2000
>>
>> Priority: N/A, Version: 6.0, Identifier: 3
>>
>> Trustpoint: N/A
>>
>> Call Manager: 10.12.200.21, Port Number: 2000
>>
>> Priority: N/A, Version: 6.0, Identifier: 2
>>
>> Trustpoint: N/A
>>
>> Call Manager: 10.12.200.22, Port Number: 2000
>>
>> Priority: N/A, Version: 6.0, Identifier: 1
>>
>> Trustpoint: N/A
>>
>> Conferencing Oper State: ACTIVE - Cause Code: NONE
>>
>> Active Call Manager: 10.12.202.1, Port Number: 2000
>>
>> TCP Link Status: CONNECTED, Profile Identifier: 2
>>
>> Reported Max Streams:

Re: [OSL | CCIE_Voice] Cbarge and SRST (Again) (Again) and (Again) what am I missing???

[cisco voip]

That bug is for srst mode auto provision none.. for provision all, it should
work

The problem you are facing of having cbarge for split second is because you
had single button cbarge when phones were registered to CUCM, disable that
setting and make it normal cbarge, they will start to work in srst mode as
well



On Fri, Aug 6, 2010 at 5:05 PM, Ashar Siddiqui  wrote:

>  I am glad that the solution proposed by Cisco is exactly what I did
> months back after trying different solutions.
>
>
>
> Ash.
>
>
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *CCIE Voice GMAIL
> *Sent:* 06 August 2010 03:13
>
> *To:* ccie_voice@onlinestudylist.com
> *Subject:* Re: [OSL | CCIE_Voice] Cbarge and SRST (Again) (Again) and
> (Again) what am I missing???
>
>
>
> I thought I’d share this with everyone as this have been extremely
> frustrating for me.  Apparently this is a known bug (well…recently known).
>
>
>
> CSCti11843
>
>
>
>
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *MARSHALL, JODY C
> (ATTBCS)
> *Sent:* Wednesday, August 04, 2010 4:55 AM
> *To:* ccie_voice@onlinestudylist.com
> *Subject:* [OSL | CCIE_Voice] Cbarge and SRST (Again) (Again) and (Again)
> what am I missing???
>
>
>
> I have read (several) post on this and have tested several different ways.
> None of which have I been able to make work. Can you please take a look and
> see if I am missing something. The first configuration is from
> auto-provision all. I had the phones registers then unregister bounced the
> router and register again. Cbarge does not work. I see the remote-in-use
> state for a second then disappears. I then registered the phones to CUCM and
> removed telephony-service reloaded the router and reconfigured
> telephony-service with auto-provision none with the second configuration
> posted. Cbarge does not work.
>
> 124-20.T5.bin
>
> telephony-service
>
>  sdspfarm units 5
>
>  sdspfarm tag 2 sitebcfb
>
>  conference hardware
>
>  srst mode auto-provision all
>
>  srst ephone template 1
>
>  srst ephone description SRST : Aug 03 2010 13:28:28 : Aug 03 2010 21:20:20
>
>  srst dn template 1
>
>  srst dn line-mode octo
>
>  max-ephones 4
>
>  max-dn 30 preference 3
>
>  ip source-address 10.12.202.1 port 2000
>
>  system message CCIEVOICE
>
>  time-zone 8
>
>  date-format dd-mm-yy
>
>  voicemail 2220
>
>  max-conferences 8 gain -6
>
>  web admin system name administrator password ccievoice
>
>  transfer-system full-consult
>
>  transfer-pattern .T
>
>  create cnf-files version-stamp 7960 Aug 03 2010 21:20:26
>
> !
>
>
>
> R2#sho sccp
>
> SCCP Admin State: UP
>
> Gateway IP Address: 10.12.202.1, Port Number: 2000
>
> IP Precedence: 5
>
> User Masked Codec list: None
>
> Call Manager: 10.12.202.1, Port Number: 2000
>
> Priority: N/A, Version: 6.0, Identifier: 3
>
> Trustpoint: N/A
>
> Call Manager: 10.12.200.21, Port Number: 2000
>
> Priority: N/A, Version: 6.0, Identifier: 2
>
> Trustpoint: N/A
>
> Call Manager: 10.12.200.22, Port Number: 2000
>
> Priority: N/A, Version: 6.0, Identifier: 1
>
> Trustpoint: N/A
>
> Conferencing Oper State: ACTIVE - Cause Code: NONE
>
> Active Call Manager: 10.12.202.1, Port Number: 2000
>
> TCP Link Status: CONNECTED, Profile Identifier: 2
>
> Reported Max Streams: 8, Reported Max OOS Streams: 0
>
> Supported Codec: g711ulaw, Maximum Packetization Period: 30
>
> Supported Codec: g711alaw, Maximum Packetization Period: 30
>
> Supported Codec: g729ar8, Maximum Packetization Period: 60
>
> Supported Codec: g729abr8, Maximum Packetization Period: 60
>
> Supported Codec: g729r8, Maximum Packetization Period: 60
>
> Supported Codec: g729br8, Maximum Packetization Period: 60
>
> Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
>
> Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
>
> Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization
> Period: 30
>
> R2#sho ephone
>
> ephone-1[0] Mac:001A.6D57.021D TCP socket:[1] activeLine:0 REGISTERED in
> SCCP ver 17/9
>
> mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
> caps:8 privacy:0
>
> IP:10.10.255.156 35781 7961  keepalive 82 max_line 6 available_line 5
>
> button 1: dn 1  number 3001 CH1   IDLE CH2   IDLE CH3
> IDLE CH4   IDLE CH5   IDLE CH6   IDLE CH7
> IDLE CH8   IDLE
>
> button 2: dn 2  number 3012 CH1   IDLE CH2   IDLE CH3
> IDLE CH4   IDLE CH5   IDLE CH6   IDLE CH7
> IDLE CH8   IDLE shared
>
> privacy button is enabled
>
> Preferred Codec: g711ulaw
>
>
>
> ephone-2[1] Mac:0019.56A3.A0D8 TCP socket:[2] activeLine:0 REGISTERED in
> S

Re: [OSL | CCIE_Voice] Supplementary Services with Remote GK

[cisco voip]

As per srnd.. there are some three commands required

emptycapabilityset
h225 connect pass thru

and 1 more which i could not recollect right now

On Tue, Aug 3, 2010 at 6:50 AM, CCIE Voice GMAIL <
givemeccievoice2...@gmail.com> wrote:

>  I have a transcoder configured and associated with both the phones and
> the GK/CUBE.  Will that do the trick?
>
>
>
> *From:* Mike Thompson [mailto:mthompson...@gmail.com]
> *Sent:* Monday, August 02, 2010 5:56 PM
> *To:* CCIE Voice GMAIL
> *Cc:* OSL Group
> *Subject:* Re: [OSL | CCIE_Voice] Supplementary Services with Remote GK
>
>
>
> Try integrating an MTP
>
> Sent from my phone, apologies for any typos.
>
>
> On Aug 2, 2010, at 8:52 PM, "CCIE Voice GMAIL" <
> givemeccievoice2...@gmail.com> wrote:
>
>  Hi everyone,
>
>
>
> Is there something special you need to configure with a Remote GK in order
> to enable transfers or holds?  I am fooling around with the various commands
> under voice service voip, but nothing is giving me any success yet.
>
>
>
> Thanks.
>
>
>
>
>
>  ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] Called and Calling numbering type

[cisco voip]

Hey Brian,

I understand your point, but what if the question says set the called number
and calling number type correctly for all the calls.

My understanding says if i am going out of local gw dialing an international
number, my calling numbering type will still be subscriber. But i always saw
people doing it international for both called and calling.

On Sat, Jul 24, 2010 at 11:02 PM, cisco voip wrote:

> Thanks Mate
>
>
> On Sat, Jul 24, 2010 at 10:35 PM, Brian Valentine 
> wrote:
>
>> You could.  It depends.  You earn points in the lab exam by
>> accomplishing the tasks that they have specifically asked you to
>> perform.  There are no "extra credit" points for making things look
>> pretty or consistent.  So, don't waste your valuable time doing things
>> that you were not asked to do.  If the question says that EVERY time
>> the call goes out BR1 to a particular destination, you should mark the
>> calling number type as X, then, sure, make it X.  If it doesn't say,
>> then they don't care - not one bit.
>>
>> You never know - they might even ask you to do something strange
>> like... set the calling party type as International on a local call
>> and set it as Subscriber on the TEHO call.  Why would they do that?
>> Because they are not testing to see if you know and can perform best
>> practices.  They want to know that you can make it do what they have
>> asked.  So, just make it do what they ask.  Get the points and then
>> move on to another task.
>>
>> What I'm saying is... "should be" is whatever the exam asks you to make
>> it.
>>
>>
>>
>> On Sat, Jul 24, 2010 at 12:40 PM, cisco voip 
>> wrote:
>> > I was planning to set calling number type using Calling party
>> transformation
>> > pattern at one go...
>> >
>> > No??
>> >
>> > On Sat, Jul 24, 2010 at 10:03 PM, Brian Valentine <
>> bkvalent...@gmail.com>
>> > wrote:
>> >>
>> >> Just make it do what the lab says.  Otherwise, don't change it.
>> >>
>> >> On Jul 24, 2010 12:32 PM, "cisco voip" 
>> wrote:
>> >>
>> >> Hello Experts,
>> >>
>> >> i am really confused with what should be the called/calling number type
>> in
>> >> case of teho calls. My guess if HQ Phone doing teho thru BR1, called
>> >> numbering type should be subscriber and calling number type should be
>> >> national..
>> >>
>> >> Need suggestions
>> >>
>> >> ___
>> >> For more information regarding industry leading CCIE Lab training,
>> please
>> >> visit www.ipexpert.com
>> >>
>> >
>> >
>>
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] Called and Calling numbering type

[cisco voip]

Thanks Mate

On Sat, Jul 24, 2010 at 10:35 PM, Brian Valentine wrote:

> You could.  It depends.  You earn points in the lab exam by
> accomplishing the tasks that they have specifically asked you to
> perform.  There are no "extra credit" points for making things look
> pretty or consistent.  So, don't waste your valuable time doing things
> that you were not asked to do.  If the question says that EVERY time
> the call goes out BR1 to a particular destination, you should mark the
> calling number type as X, then, sure, make it X.  If it doesn't say,
> then they don't care - not one bit.
>
> You never know - they might even ask you to do something strange
> like... set the calling party type as International on a local call
> and set it as Subscriber on the TEHO call.  Why would they do that?
> Because they are not testing to see if you know and can perform best
> practices.  They want to know that you can make it do what they have
> asked.  So, just make it do what they ask.  Get the points and then
> move on to another task.
>
> What I'm saying is... "should be" is whatever the exam asks you to make it.
>
>
>
> On Sat, Jul 24, 2010 at 12:40 PM, cisco voip 
> wrote:
> > I was planning to set calling number type using Calling party
> transformation
> > pattern at one go...
> >
> > No??
> >
> > On Sat, Jul 24, 2010 at 10:03 PM, Brian Valentine  >
> > wrote:
> >>
> >> Just make it do what the lab says.  Otherwise, don't change it.
> >>
> >> On Jul 24, 2010 12:32 PM, "cisco voip" 
> wrote:
> >>
> >> Hello Experts,
> >>
> >> i am really confused with what should be the called/calling number type
> in
> >> case of teho calls. My guess if HQ Phone doing teho thru BR1, called
> >> numbering type should be subscriber and calling number type should be
> >> national..
> >>
> >> Need suggestions
> >>
> >> ___
> >> For more information regarding industry leading CCIE Lab training,
> please
> >> visit www.ipexpert.com
> >>
> >
> >
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] Called and Calling numbering type

[cisco voip]

I was planning to set calling number type using Calling party transformation
pattern at one go...

No??

On Sat, Jul 24, 2010 at 10:03 PM, Brian Valentine wrote:

> Just make it do what the lab says.  Otherwise, don't change it.
>
> On Jul 24, 2010 12:32 PM, "cisco voip"  wrote:
>
> Hello Experts,
>
> i am really confused with what should be the called/calling number type in
> case of teho calls. My guess if HQ Phone doing teho thru BR1, called
> numbering type should be subscriber and calling number type should be
> national..
>
> Need suggestions
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

[OSL | CCIE_Voice] Called and Calling numbering type

[cisco voip]

Hello Experts,

i am really confused with what should be the called/calling number type in
case of teho calls. My guess if HQ Phone doing teho thru BR1, called
numbering type should be subscriber and calling number type should be
national..

Need suggestions
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] NTP

[cisco voip]

But "ntp master" command is necessary for the router to be able to be ntp
server for anyone else. i agree about stratum chain, but without ntp master
command publisher should not be able to get its clock.

you can check show ntp associations on the router, after ntp master command
only it will show you the local interface.

now if you want to use the stratum to be 1 more from whatever the stratum is
on actual source, you do not need stratum command, for anything else you
need that commnd

On Fri, Jul 9, 2010 at 4:42 AM, Mark Holloway  wrote:

> Cool, thanks Graham and Randall.
>
> On Jul 8, 2010, at 4:11 PM, Graham Hopkins wrote:
>
> > Default stratum is 8 so a simple ntp master will work
> >
> > Graham
> >
> > On 8 Jul 2010, at 23:59, Mark Holloway  wrote:
> >
> >> Yikes, I meant "ntp master stratum X" not "ntp server stratum X"
> >>
> >> On Jul 8, 2010, at 3:57 PM, Mark Holloway wrote:
> >>
> >>> If a router (for example, HQ) is configured with the "ntp server
> x.x.x.x" command to sync time from another source, but I want another device
> (such as PUB) to get its time from the HQ router, do I also need to
> configure the HQ router with "ntp server stratum X" or can UCM simply get
> the time sync from HQ without the stratum command?
> >>> ___
> >>> For more information regarding industry leading CCIE Lab training,
> please visit www.ipexpert.com
> >>
> >> ___
> >> For more information regarding industry leading CCIE Lab training,
> please visit www.ipexpert.com
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] Presence call-list in CME

[cisco voip]

did that

On Sun, Jul 4, 2010 at 8:28 AM, Shadow of Voice
wrote:

> use presence-call list under ephone and under ephone-dn use allow watch
>
>
>
> On Sat, Jul 3, 2010 at 10:29 PM, cisco voip wrote:
>
>> Hello Experts,
>>
>> Is there some gotcha with presence call list in CME. It seems i did
>> everything correctly, status is show in blf-speeddial button but not is the
>> directory>received call.
>>
>> Is there an extra command to enable it in directories.
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>
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For more information regarding industry leading CCIE Lab training, please visit 
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[OSL | CCIE_Voice] Presence call-list in CME

[cisco voip]

Hello Experts,

Is there some gotcha with presence call list in CME. It seems i did
everything correctly, status is show in blf-speeddial button but not is the
directory>received call.

Is there an extra command to enable it in directories.
___
For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] Connected number display

[cisco voip]

As soon as i thought that i understood the dial plan  and there is a
different  thing.
Nice idea Daniel, I remember doing something on called party x'formation
pattern and display did change. I don't remember the exact scenarion, just
remember it din't work at first and i restarted the CCM Service and display
changed.

i will test it in lab tomorrow.

On Tue, Jun 22, 2010 at 2:12 PM, Angel Perez  wrote:

>  Hi:
>
> Imho this won't work, I've tested yesterday and phone display has to be
> manipulated on rp not rl, I haven't test it at called party xformation level
> but  Daniel's aproach seems to be the only working solution
>
> Thanks
>
> --
> Date: Mon, 21 Jun 2010 23:58:41 -0700
> From: ciscovoiceg...@gmail.com
>
> To: ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] Connected number display
>
> Daniel,
>
> If you do not adjust the called number display on the route pattern, the
> called number display settings on the route list will go into effect.  Have
> you tried to manipulate the called number on the route list?
>
>
>  *Matthew Berry*
>
> *A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written*
>
>
>
> *Vitals:*
>
> *GVoice: *+1.612.424.5044
>
> *Gmail*: ciscovoiceg...@gmail.com
>
> *Skype*: ciscovoiceguru
>
> *Twitter*: ciscovoiceguru
>
> On 6/21/2010 11:26 PM, Daniel Berlinski wrote:
>
> Manipulation at the route list level does not affect how the dialed number
> is updated on the phone display.
>
> I read this as per below:
> "If I have two MGCP gateways BR1 and BR2, and call should go thru BR1 and
> if it fails it should go thru BR2.
> Requirement is if call goes through BR1, called number on my display should
> be 7 digits. If it goes thru BR2, called number should be 10 digits."
>
> How would manipulation at the route list help in this scenario?
>
> I have just tested here by manipulating the dialed number at the route
> pattern for the first choice gateway (MGCP BR1 - 7Digits) and by using
> called party xformation pattern for the second choice gateway (MGCP-BR2)  In
> my case I could not do it for 10 digits because my BR2 router is in Spain.
> The phone display updates as per both transformation configs.
>
> If this is not correct please let me know what I'm missing
> Cheers
>
> On Tue, Jun 22, 2010 at 2:20 PM, Berry, Matthew J. <
> mjbe...@krollontrack.com> wrote:
>
> Daniel,
>
> You best bet would be to do the manipulation at the route list level for
> such a request.
> - Sent from my Blackberry
>
>  --
> *From*: ccie_voice-boun...@onlinestudylist.com <
> ccie_voice-boun...@onlinestudylist.com>
> *To*: Angel Perez 
> *Cc*: osl osl 
> *Sent*: Mon Jun 21 16:04:44 2010
>
> *Subject*: Re: [OSL | CCIE_Voice] Connected number display
>
>  Hello Guys
>
> Just an idea and please ignore if this is a silly one or let me know if you
> have already tested this.
>
> Could you try to have your manipulation done at route pattern level for BR1
> and for BR2 add a called party xformation in order to update the phone
> display when BR1 is down?  As far as my understanding goes ANI manipulations
> at route pattern and (DNIS) called party transformation patterns applied to
> egress gateways will also have the cosmetic effect to phones screens.
>
> I will give this a go as soon as I have access to equipment again and will
> update
>
> Best Regards
> Daniel
>
>
>
>
>
> On Mon, Jun 21, 2010 at 11:13 PM, Angel Perez  wrote:
>
> Yes you are right, tested today, ccm engine will not try with another route
> pattern although controller/gw associated to the first rp is not "up". I
> thought ccm would follow the same behaviour as a h323 gw.
>
> Since the only way I know to change phone display number is through route
> patt, my conclusion is that your requirements are not possible to be
> satified...
>
> Is this an exercise from a workbook or something you want to test? In case
> it's the first one let us know the solution becouse I can't think a way to
> make this work with ucm only.
>
> Thanks
>
> --
> Date: Sun, 20 Jun 2010 17:28:59 +0530
>
> Subject: Re: [OSL | CCIE_Voice] Connected number display
> From: voip.ccieci...@gmail.com
> To: gorr...@hotmail.com
> CC: siddas...@gmail.com; ccie_voice@onlinestudylist.com
>
>
> i tested bot the RP first.. then i did a no mgcp command on GW1
>
> On Sun, Jun 20, 2010 at 4:52 PM, Angel Perez  wrote:
>
> Hi:
>
> Did you test both  rp alone first to make sure it working correctly?
>
> Did you shutdo

Re: [OSL | CCIE_Voice] Connected number display

[cisco voip]

i tested bot the RP first.. then i did a no mgcp command on GW1

On Sun, Jun 20, 2010 at 4:52 PM, Angel Perez  wrote:

>  Hi:
>
> Did you test both  rp alone first to make sure it working correctly?
>
> Did you shutdown controller at br1 before testing backup path?
>
> thx
>
> --
> Date: Sun, 20 Jun 2010 11:49:27 +0100
> From: siddas...@gmail.com
> To: voip.ccieci...@gmail.com
> CC: gorr...@hotmail.com; ccie_voice@onlinestudylist.com
>
> Subject: Re: [OSL | CCIE_Voice] Connected number display
>
>
> Did you also try what I suggested? masking Called party at RL detail level!
>
> cisco voip wrote:
>
> I tried this just now. and it is not working,
>
> So what i was thinking is correct, it can match only one route pattern and
> call cannot come back.
>
> Is there any other way anyone would think of??
>
>
>
> On Sun, Jun 20, 2010 at 12:00 AM, Angel Perez  wrote:
>
> Hi Ash, I think that to change  calling number at phone display you may do
> transformation at rp level, correct me if i'm wrong
>
> thx
>
> --
> Date: Sat, 19 Jun 2010 12:34:08 +0100
> From: siddas...@gmail.com
> To: gorr...@hotmail.com
> CC: voip.ccieci...@gmail.com; ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] Connected number display
>
>
> Sorry Ignore my last post, I thought you are asking about Calling party
> number (ANI).
> The one Angel mentioned is a possible solution or try this one...make one
> route pattern, Create two RG in the RL, then place mask under Called party
> like XXX and XX under Route list detail level. I have not tested
> it so give it a try and let us know how it works.
>
> Ash>
>
> Angel Perez wrote:
>
> Hi:
>
> The only way I can imagine to make this work is with to different route
> patterns, instead with one route pattern and a route list with two options,
> something like this:
>
> rp1:  91[2-9]XX.[2-9]XX  DDI PREDOT, PT=br1-local-first-option
> rp2:  91.[2-9]XX[2-9]XX  DDI PREDOT, PT=br1-local-sec-option
>
> br1 phone 1: css (phones,911,br1-local-first-option, br1-local-sec-option,
> ld, ...)
>
> Becouse rp1 and rp2 are and equal match for UCM call processing engine, the
> pt orther will be the tie breaker, so the first choice would be rp1, and
> second choice would be rp2.
>
> Let us know how it goes
>
> Regards
> --
> Date: Sat, 19 Jun 2010 16:01:09 +0530
> From: voip.ccieci...@gmail.com
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] Connected number display
>
> Hi Experts,
>
> If I have two MGCP gateways BR1 and BR2, and call should go thru BR1 and if
> it fails it should go thru BR2.
> Requirement is if call goes through BR1, called number on my display should
> be 7 digits. If it goes thru BR2, called number should be 10 digits.
>
> From what i understand, display number is the manipulated number in Route
> Pattern. So I am not really sure how to change the display number on the
> basis of what gateway call is going out.
> Any Suggestions?
>
> --
> Hotmail: Trusted email with powerful SPAM protection. Sign up 
> now.<https://signup.live.com/signup.aspx?id=60969>
>
> --
>
> ___
> For more information regarding industry leading CCIE Lab training, please 
> visit www.ipexpert.com
>
>
>
>  --
> Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up
> now. <https://signup.live.com/signup.aspx?id=60969>
>
>
>
>
> --
> Hotmail: Free, trusted and rich email service. Get it 
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>
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Re: [OSL | CCIE_Voice] Connected number display

[cisco voip]

I tried this just now. and it is not working,

So what i was thinking is correct, it can match only one route pattern and
call cannot come back.

Is there any other way anyone would think of??



On Sun, Jun 20, 2010 at 12:00 AM, Angel Perez  wrote:

>  Hi Ash, I think that to change  calling number at phone display you may do
> transformation at rp level, correct me if i'm wrong
>
> thx
>
> --
> Date: Sat, 19 Jun 2010 12:34:08 +0100
> From: siddas...@gmail.com
> To: gorr...@hotmail.com
> CC: voip.ccieci...@gmail.com; ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] Connected number display
>
>
> Sorry Ignore my last post, I thought you are asking about Calling party
> number (ANI).
> The one Angel mentioned is a possible solution or try this one...make one
> route pattern, Create two RG in the RL, then place mask under Called party
> like XXX and XX under Route list detail level. I have not tested
> it so give it a try and let us know how it works.
>
> Ash>
>
> Angel Perez wrote:
>
> Hi:
>
> The only way I can imagine to make this work is with to different route
> patterns, instead with one route pattern and a route list with two options,
> something like this:
>
> rp1:  91[2-9]XX.[2-9]XX  DDI PREDOT, PT=br1-local-first-option
> rp2:  91.[2-9]XX[2-9]XX  DDI PREDOT, PT=br1-local-sec-option
>
> br1 phone 1: css (phones,911,br1-local-first-option, br1-local-sec-option,
> ld, ...)
>
> Becouse rp1 and rp2 are and equal match for UCM call processing engine, the
> pt orther will be the tie breaker, so the first choice would be rp1, and
> second choice would be rp2.
>
> Let us know how it goes
>
> Regards
> --
> Date: Sat, 19 Jun 2010 16:01:09 +0530
> From: voip.ccieci...@gmail.com
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] Connected number display
>
> Hi Experts,
>
> If I have two MGCP gateways BR1 and BR2, and call should go thru BR1 and if
> it fails it should go thru BR2.
> Requirement is if call goes through BR1, called number on my display should
> be 7 digits. If it goes thru BR2, called number should be 10 digits.
>
> From what i understand, display number is the manipulated number in Route
> Pattern. So I am not really sure how to change the display number on the
> basis of what gateway call is going out.
> Any Suggestions?
>
> --
> Hotmail: Trusted email with powerful SPAM protection. Sign up 
> now.
>
> --
>
> ___
> For more information regarding industry leading CCIE Lab training, please 
> visit www.ipexpert.com
>
>
>
> --
> Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up
> now. 
>
___
For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] Connected number display

[cisco voip]

Hi All,

Thanks for quick response.
>From what i understand, it matches a route pattern and call will go out. if
it fails, it will try second gateway from route group but i really don't
think it can match another route pattern.
I may be wrong, will test this and let you know.


On Sat, Jun 19, 2010 at 5:49 PM, Shadow of Voice
wrote:

> Agree with angel and Ash last post 
>
> On Sat, Jun 19, 2010 at 7:34 AM, Ashar Siddiqui wrote:
>
>> Sorry Ignore my last post, I thought you are asking about Calling party
>> number (ANI).
>> The one Angel mentioned is a possible solution or try this one...make one
>> route pattern, Create two RG in the RL, then place mask under Called party
>> like XXX and XX under Route list detail level. I have not tested
>> it so give it a try and let us know how it works.
>>
>> Ash>
>>
>> Angel Perez wrote:
>>
>>  Hi:
>>
>> The only way I can imagine to make this work is with to different route
>> patterns, instead with one route pattern and a route list with two options,
>> something like this:
>>
>> rp1:  91[2-9]XX.[2-9]XX  DDI PREDOT, PT=br1-local-first-option
>> rp2:  91.[2-9]XX[2-9]XX  DDI PREDOT, PT=br1-local-sec-option
>>
>> br1 phone 1: css (phones,911,br1-local-first-option, br1-local-sec-option,
>> ld, ...)
>>
>> Becouse rp1 and rp2 are and equal match for UCM call processing engine,
>> the pt orther will be the tie breaker, so the first choice would be rp1, and
>> second choice would be rp2.
>>
>> Let us know how it goes
>>
>> Regards
>> --
>> Date: Sat, 19 Jun 2010 16:01:09 +0530
>> From: voip.ccieci...@gmail.com
>> To: ccie_voice@onlinestudylist.com
>> Subject: [OSL | CCIE_Voice] Connected number display
>>
>> Hi Experts,
>>
>> If I have two MGCP gateways BR1 and BR2, and call should go thru BR1 and
>> if it fails it should go thru BR2.
>> Requirement is if call goes through BR1, called number on my display
>> should be 7 digits. If it goes thru BR2, called number should be 10 digits.
>>
>> From what i understand, display number is the manipulated number in Route
>> Pattern. So I am not really sure how to change the display number on the
>> basis of what gateway call is going out.
>> Any Suggestions?
>>
>> --
>> Hotmail: Trusted email with powerful SPAM protection. Sign up 
>> now.
>>
>> --
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please 
>> visit www.ipexpert.com
>>
>>
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
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[OSL | CCIE_Voice] Connected number display

[cisco voip]

Hi Experts,

If I have two MGCP gateways BR1 and BR2, and call should go thru BR1 and if
it fails it should go thru BR2.
Requirement is if call goes through BR1, called number on my display should
be 7 digits. If it goes thru BR2, called number should be 10 digits.

>From what i understand, display number is the manipulated number in Route
Pattern. So I am not really sure how to change the display number on the
basis of what gateway call is going out.
Any Suggestions?
___
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www.ipexpert.com

Re: [OSL | CCIE_Voice] Called Party Transformation

[cisco voip]

Hi Amy,

Thanks a lot for quick reply.
I tried this scenario in the lab, and it is working as you mentioned for
MGCP gateways.

But for H323 gateway, the display on the phone is the destination-pattern of
outgoing dial-peer, is there a way to manipulate that too.


On Thu, Jun 17, 2010 at 8:36 PM, Amy Ryan  wrote:

>
> When utilizing the called party transformation pattern CSS to manipulate
> digits when conducting outbound calls you are modifying the dialed digits
> (DNIS) that are presented to the destination (in this case the PSTN) and
> those same digits will be reflected on your display.  If you have decided
> to do called party # manipulation at the gateway via a Called Party
> Transformation Pattern CSS (that see a matching pattern) it will always
> override anything done at the RP/RL display or otherwise.  If you run a
> “debug isdn q931” on the gateway you will see this.
>
> If you are trying to force the phone display of the phone dialing 41031000
> to show 08041031000 on the display only, you could achieve this by doing
> Called Party # Transformations at both Route Pattern (RP) and Route List
> (RL).
>
> If you perform Called # Transformations at both RP and RL, the called #
> transformations done at the RP level will then affect display on calling
> phone and the called # transformations done at the RL level will affect
> Called # sent to gateway so that the PSTN accepts the format and processes
> the call.
>
> So in this case you may have both the following:
>
> *Route Pattern
> *Pattern = 9.41031000
> Called # Transformation = DDI Predot, Prefix 080
>
> *Route List (within route group)
> *Called # Transformation = DDI Predot
>
> This will invoke 08041031000 to show on the phone display and 41031000 to
> be sent to the PSTN.
>
>
> *In summary**, *digit manipulation can be done all in RP or all in RL.
>  The RL will always override the RP when determining what is sent as dialed
> digits to destination.  However when doing called party # transformations at
> this level, and utilizing both RL and RP, manipulations done at RP will
> always affect display of calling phone and the manipulation done at the RL
> will allow for appropriate digits to be send to destination.
>
> HTH,
> Amy
>
>
>
> ---
> Amy Ryan – CCIE #24677 (Voice)
> Technical Instructor - IPexpert, Inc.
> Mailto: *ar...@ipexpert.com
> *Telephone: +1.810.326.1444
> Live Assistance, Please visit: www.ipexpert.com/chat <*
> http://www.ipexpert.com/chat*>
> eFax: +1.810.454.0130
>
> IPexpert is a premier provider of Self-Study Workbooks, Video on Demand,
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> our public website at www.ipexpert.com <*http://www.ipexpert.com/*>
>
>
>
> --
> *From: *cisco voip 
> *Date: *Thu, 17 Jun 2010 19:12:29 +0530
> *To: *osl osl 
> *Subject: *[OSL | CCIE_Voice] Called Party Transformation
>
>
> Hi List,
>
> I am trying to understand the use of
> Call Routing > Transformation Pattern > Called/Calling
>
> From what i understand Calling party transformation pattern can be used for
> localization by displaying 7 digits while ringing.
>
> Based on this i thought i can use called party transformation mask while
> calling outside.
> I used called party transformation CSS on outbound gateway and add a called
> party Xform pattern as
> 4103 -> Predot prefix digits- 080
>
> i thought, if i dial 41031000 number that will go to gateway is 41031000
> but on calling phone, number will be displayed as 08041031000,
>
> But CUCM is behaving weird, and it gives me different results everytime i
> start CCM service.
>
> Any idea or documentation on this?
>
>
>
> --
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
___
For more information regarding industry leading CCIE Lab training, please visit 
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[OSL | CCIE_Voice] Called Party Transformation

[cisco voip]

Hi List,

I am trying to understand the use of
Call Routing > Transformation Pattern > Called/Calling

>From what i understand Calling party transformation pattern can be used for
localization by displaying 7 digits while ringing.

Based on this i thought i can use called party transformation mask while
calling outside.
I used called party transformation CSS on outbound gateway and add a called
party Xform pattern as
4103 -> Predot prefix digits- 080

i thought, if i dial 41031000 number that will go to gateway is 41031000 but
on calling phone, number will be displayed as 08041031000,

But CUCM is behaving weird, and it gives me different results everytime i
start CCM service.

Any idea or documentation on this?
___
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Re: [OSL | CCIE_Voice] External Phone # mask in Unity

[cisco voip]

You can use CTI Route point

On Thu, May 27, 2010 at 8:22 AM, Salman Shaikh wrote:

> Is there any way to assign " external phone number mask" to voicemail
> pilot number (in CUE and UC both)? on display when you call from PSTN VM
> Pilot number 
> I appreciate your advise.
>
> Thanks
> sal
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
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Re: [OSL | CCIE_Voice] CME-CUE VOICEVIEW

[cisco voip]

Yeah
http://cue/voiceview/authentication/authenticate.do

is correct


On Thu, May 20, 2010 at 6:51 PM, Angel Perez  wrote:

>  Hi:
>
> Wich is the correct one?
>
> This one?:
>
> url authentication http://cue/voiceview/authentication/authenticate.do
>
> thx
>
> --
> Date: Thu, 20 May 2010 18:01:55 +0530
> Subject: Re: [OSL | CCIE_Voice] CME-CUE VOICEVIEW
> From: voip.ccieci...@gmail.com
> To: gorr...@hotmail.com
>
>
> You have wrong authentication URL
>
> On Thu, May 20, 2010 at 4:51 PM, Angel Perez  wrote:
>
> Hi all:
>
> I've the following problem with voiceview and CME:
>
> I've sucsesfully configure the voiceview service for phones, I access the
> service (no pin asked) but once I see the menu options (1 inbox, 2 Send
> Messages, 3 etc) I can't select any of the options neither logout with
> Logout button, I've follow these steps but with no luck
>
>
> http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/rel3_2/administrator/voicemail/7vview.html
>
> Here is my config per the above guide:
> **
> *CME:*
> **
> *telephony-service*
> * url services 
> **http:///voiceview/common/login.do*
> *
>  url authentication 
> **http://cme-ip-address/CCMCIP/authenticate.asp*
> *  *
> * authentication credential Admin cisco*
>
> *CUE:*
> **
> site name local
>  phone-authentication Admin cisco
>  end site
> **
> *cue# show voiceview configuration
> Phone service URL:   **http:///voiceview/common/login.do
> * 
> *Enabled: Yes
> Idle Timeout (minutes):  5*
> **
> **
> Am I missing something?
>
> Regards
>
>
> --
> Hotmail: Free, trusted and rich email service. Get it 
> now.
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>
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