Re: [OSL | CCIE_Voice] UCCE Script: Transfer Call to IP Phone UCCE Script

[Pavan K]

As for the original question, it should be possible to return a label to
whatever target on UCM in the ICM script. Am I missing something non
trivial?
 On Apr 6, 2014 1:19 PM, "Chrysostomos Christofi" 
wrote:

>  Hi
>
>
>
> If you have CUE you can achieve this task in the same way as into UCCX
>
>
>
> #Chrysostomos
>
>
>
>
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *CCIEing
> *Sent:* Sunday, April 6, 2014 9:28 AM
> *To:* ccie_voice@onlinestudylist.com
> *Subject:* [OSL | CCIE_Voice] UCCE Script: Transfer Call to IP Phone UCCE
> Script
>
>
>
> Dear Group Members,
>
>
>
> I would to ask about this feature ..After a time check (if the time after
> mid night) I have a UCCE system and I need to create a script that allow
> the call to be transferred to an IP Phone (Not Agent) just a number or PSTN
> number , the most important is "This is not agent".
>
>
>
> As you may all know this is easy from the UCCX , but is that doable from
> UCCE ?
>
>
>
> Thanks
>
>
>
> ___
> Free CCIE R&S, Collaboration, Data Center, Wireless & Security Videos ::
>
> iPexpert on YouTube: www.youtube.com/ipexpertinc
>
___
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Re: [OSL | CCIE_Voice] Disabling almost all mailboxes

[Pavan K]

Here is a simpler solution.
Create a new partition on unity and move all the users you want disabled
there. Don't add it to any search space and you are done
On Mar 6, 2014 4:15 PM, "Christian Holst"  wrote:

> If people are not able to leave voice messages (no forward) then MWI won't
> be turned on.
> If it's on already people should listen to their voicemails :) - which
> will turn it off
>
> In the odd case it's on by accident force it off.
>
> regards
> Christian Holst
>
>
> -Original Message-
> From: Isamar Maia [mailto:isa...@gmail.com]
> Sent: 6. marts 2014 22:39
> To: Christian Holst
> Cc: ccie_voice@onlinestudylist.com; mauri...@imtech.com.br
> Subject: Re: [OSL | CCIE_Voice] Disabling almost all mailboxes
>
> Is it gonna disable the MWI led as well ?
>
> Isamar
>
>
> 2014-03-06 17:52 GMT-03:00 Christian Holst :
> > If just disable, i'd remove the forward and leave the unity users
> >
> > If you delete the users all calls will end up in default greeting -
> can't imagine a customer would be happy about that one.
> >
> > Regards
> > Christian Holst
> > System Engineer UC
> > CCIE Voice #41370
> >
> >
> > -Original Message-
> > From: ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] On Behalf Of Isamar Maia
> > Sent: 6. marts 2014 21:32
> > To: ccie_voice@onlinestudylist.com
> > Subject: [OSL | CCIE_Voice] Disabling almost all mailboxes
> >
> > Hi folks,
> >
> > What is the best ways to disable most of mailboxes on Unity, leaving
> just some of them unchanged ? Removing the users on Unity admin GUI ?
> >
> >
> >
> > --
> > Isamar Maia
> > Cel. VIVO SSA:  (55) 71-9940-2012
> > Cel. TIM   SSA:  (55) 71-9289-5128
> > Cel. Claro SSA:  (55) 71-9146-8575
> > Fixo:  (55) 71-4062-8688
> > Skype ID: isamar.maia
> > "A vida é muito curta para ser pequena" (Benjamin Disraeli)
> ___
> > Free CCIE R&S, Collaboration, Data Center, Wireless & Security Videos ::
> >
> > iPexpert on YouTube: www.youtube.com/ipexpertinc
>
>
>
> --
> Isamar Maia
> Cel. VIVO SSA:  (55) 71-9940-2012
> Cel. TIM   SSA:  (55) 71-9289-5128
> Cel. Claro SSA:  (55) 71-9146-8575
> Fixo:  (55) 71-4062-8688
> Skype ID: isamar.maia
> "A vida é muito curta para ser pequena" (Benjamin Disraeli)
> ___
> Free CCIE R&S, Collaboration, Data Center, Wireless & Security Videos ::
>
> iPexpert on YouTube: www.youtube.com/ipexpertinc
>
___
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Re: [OSL | CCIE_Voice] CTI Application Server (TCP/2748) vs JTAPI Application Server (TCP/2789)

[Pavan K]

Jtapi/tapi interfaces are northbound interfaces which are on top of the CTI
layer.

Taking the example of UCCX, UCCX can sync with ucm and download jtapi
libraries from ccm. Its built in jtapi client uses those libraries to
communicate with CTI on the ucm server.

The term rmjtapi refers to the local credentials used by its jtapi client
to connect to CTI.

Hope that helps
On Nov 2, 2013 4:08 AM, "Somphol Boonjing"  wrote:

> Could anyone help explain or refer me to the documentation that help me
> understand the role of JTAPI Application Server (tcp/2789) a bit more?   I
> am interested to learn about which application server use that particular
> port TCP/2789? (CUC / UCCX / CUE / CUPC)
>
> I know that both CUE and CUPC (Deskphone mode) and UCCX, all of them, talk
> to CTI Application Server at port TCP/2748, but does JTAPI Application
> Serer at TCP/2789 ever get used by any of those application server/client?
>
> Note: I find it very confusing when people use rmjtapi account name (in
> case of UCCX) or cuejtapi (in case of CUE) to talk to CTI Application
> Server (TCP/2748) which really is a CTI Application Server and is not JTAPI
> Application (TCP/2789).
>
> REF:
>
> http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/port/9_1_1/CUCM_BK_T2CA6EDE_00_tcp-port-usage-guide-91_chapter_01.html
>
> http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/port/8_0_1/portlist801.html
>
> http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/port/7_0/CCM_7.0PortList.pdf
>
> Cisco Unified Communications App
>
> Unified CM
>
> 2748 / TCP
>
> CTI application server
>
> Cisco Unified Communications App
>
> Unified CM
>
> 2749 / TCP
>
> TLS connection between CTI applications (JTAPI/TSP) and CTIManager
>
> Cisco Unified Communications App
>
> Unified CM
>
> 2789 / TCP
>
> JTAPI application server
>
> See Also:
>
> CUPC Port Usage -
> http://www.cisco.com/en/US/docs/voice_ip_comm/cupc/7_1/english/release/notes/cupc71.html
> UCCX Port Usage -
> http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_7_0/configuration/guide/uccx70prtuti.pdf
> CUE Integration Guide that suggests TCP/2748 is used (and there is no
> reference to TCP/2789 at all) -
> http://www.cisco.com/en/US/products/sw/voicesw/ps5520/products_configuration_example09186a0080289ef0.shtml
>
>
>
> ___
> 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] AAR and SNR

[Pavan K]

The behavior should not be any different if a shared line is configured
between two devices but only one device rings because the other does not
have sufficient bw. AAR would not kick in as a CCRej would not be sent up
by LC.
On Sep 11, 2013 2:42 PM, "Daniel Pagan"  wrote:

>  Slight correction – Location Bandwidth Manager is related to Enhanced
> Location CAC, which is not part of this discussion. What I should have said
> is simply that “Available bandwidth shows we’re requesting too much than
> what’s available” without the mention of LBM. 
>
> ** **
>
> Everything else applies.
>
> ** **
>
> Hope this helps.
>
> ** **
>
> - Dan**
>
> ** **
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Daniel Pagan
> *Sent:* Wednesday, September 11, 2013 2:57 PM
> *To:* Ovidiu Popa; ccie_voice@onlinestudylist.com
> *Subject:* Re: [OSL | CCIE_Voice] AAR and SNR
>
> ** **
>
> Confirmed – this appears to be standard behavior. Recreating this in the
> lab and reviewing CCM SDI traces (I’m on a 8.6 cluster) reveals a pretty
> logical explanation for this. First, I should mention the LineControl
> process is responsible for dispatching/distributing call attempts to
> devices associated with a Directory Number matched during the DA process.
> With that said, here’s what I’ve done and what I see…
>
> ** **
>
> *Scenario – Mobility is configured for PhoneB and tested to work
> successfully…*
>
> **1.   **PhoneA calls PhoneB – I’ve set the location bandwidth below
> the bitrate being negotiated to enforce AAR
>
> **2.   **LineControl lists all devices associated with this dialed
> Directory Number
>
> **a.**Included in this list is the device name of PhoneB and the
> device name of its Remote Destination Profile
>
> **3.   **For PhoneB (my desk phone) Location Bandwidth Manager shows
> that more bandwidth is being requested than it can subtract
>
> **a.**We cancel the call attempt to PhoneB
>
> **4.   **Because the Remote Destination Profile was detected as an
> associated device on the dialed Directory Number, we begin routing the call
> to the Remote Destination, DA process is successful, and the call is
> extended to the Remote Destination. This results in *no AAR* attempts
> because CUCM still has the ability to route the call to an associated
> device. This is key.
>
> ** **
>
> *In a scenario where Mobility is disabled, we see AAR being invoked
> because…*
>
> **1.   **LineControl lists all devices associated with this dialed
> Directory Number… only this time the only device listed is our desk phone.
> The RDestProfile is not listed.
>
> **2.   **Location BW Mgr shows we’re out of available bandwidth
>
> **3.   **At this point, low bandwidth was detected, we have no
> available devices to route through LineControl, so we begin using AAR as
> the last resort.
>
> ** **
>
> Hope this helps.
>
> ** **
>
> *Daniel Pagan *| CCIE-V #25689
>
> Managed Services Technical & Training Lead
>
> *+1-212-616-7843* office
> *+1-212-616-7850* fax
>
> www.fidelus.com**
>
> ** **
>
> *From:* ccie_voice-boun...@onlinestudylist.com [
> mailto:ccie_voice-boun...@onlinestudylist.com]
> *On Behalf Of *Ovidiu Popa
> *Sent:* Sunday, September 08, 2013 11:26 AM
> *To:* ccie_voice@onlinestudylist.com
> *Subject:* [OSL | CCIE_Voice] AAR and SNR
>
> ** **
>
> Hello everyone
>
> ** **
>
> Can anyone confirm the following results ?
>
> A - If Mobile Connect is enabled for a User then AAR calls will ring only
> the remote destination, the AAR to the DN call will NOT be extended
>
> B - If Mobile Connect is disabled for a User, AAR calls will ring the AAR
> target (as specified by the AAR Mask or as generated by AAR Group + prefix
> + External Phone Number Mask)
>
> ** **
>
> In the second test I do not see the CUCM invoking the AAR process so I'm
> guessing this is expected behavior.
>
> ** **
>
> Can anyone confirm my results?
>
> ** **
>
> Thanks,
>
> Ovidiu
>
> ** **
>
> ___
> 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] Generate a report for number of calls into PRI

[Pavan K]

If I am not mistaken, all perfmon counters are also logged to a CSV file by
ucm for investigation by TAC should a system issue arise. I can't remember
the name of the file but it must be in the active log directory somewhere
On Sep 4, 2013 9:59 PM, "John Boxold"  wrote:

> One option you could use the RTMT on a specific pc and create a customized
> alert and set it to log, the reports can be opened in excel.
> I have set the alarms to notify when a specific threshold is hit and send
> out an email alert for a PRI I set the limit to 19 active channels.
>
> I have used a temp license for Operations Manager and let it provide the
> graphing for your gateways, this can be set to poll automatically.
>
> It really depends on the amount of time you have available to generate,
> parse, and review the data.
>
> My personal opinion would be to let the telco provide the reporting for
> usage.
>
>
> Sent from my iPad
>
> On Sep 4, 2013, at 7:05 PM, CCIE Voice Aspirant <
> ccievoice2013.2...@gmail.com> wrote:
>
> CDR/CAR should be able to provide breakdown by PRI since it's MGCP.
>
> On Sep 4, 2013, at 5:34 PM, Edgar Feliz  wrote:
>
> TELCO can provide a usage report for each PRI, who is the SP?
>
> Edgar
>
>
> On Tue, Sep 3, 2013 at 2:23 PM, Hesham Abdelkereem <
> heshamcentr...@gmail.com> wrote:
>
>> Dear Experts,
>>
>> I have 12 PRI configured as MGCP gateways and would like to replace them
>> by a CUBE.
>>
>> Now, I would like to make Statistics/Feasability study about the number
>> of concurrent calls on each PRI for example today from 8am to 5PM.
>> Is there is anyway I can do that? That will help me in the calculation to
>> order the number of concurrent calls properly when I migrate into SIP.
>>
>>
>> Thanks,
>> Hesham
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>> Are you a CCNP or CCIE and looking for a job? Check out
>> www.PlatinumPlacement.com
>>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> Are you a CCNP or CCIE and looking for a job? Check out
> www.PlatinumPlacement.com
>
>
> ___
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> visit www.ipexpert.com
>
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>
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Re: [OSL | CCIE_Voice] Gatekeeper back up

[Pavan K]

Check your retry interval on device -> gatekeeper
On Aug 10, 2013 6:42 PM, "Karen Johnson"  wrote:

> when we shut Gatekeeper, it always take time to go back up.
> is there any command to speed it up ?
>
> K
>
> ___
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> visit www.ipexpert.com
>
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>
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Re: [OSL | CCIE_Voice] CtiGwy Application User

[Pavan K]

Ctigwy user is for desk phone control.
If you don't need cupc to work in desk phone control mode, no need for this
user.
On Jul 25, 2013 12:51 AM, "Devakanth Gangavarapu" 
wrote:

> Hi
>
> Cisco Presence solution is not integrated with JTAPI / TAPI
> It either uses SCCP or SIP
> It does not need CtiGwy application user
>
> Cheers
> Dev
>
>
> On Thu, Jul 25, 2013 at 4:58 AM, Barrera, Hugo 
> wrote:
>
>>  Hi,
>>
>> ** **
>>
>> Do you really need to add the CtiGwy Application User for the Presence
>> section? 
>>
>> ** **
>>
>> *Hugo *
>>
>> ** **
>>
>> ___
>> 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] CUPC

[Pavan K]

Its most likely a firewall blocking rtp. Cannot be routes as the signaling
is OK (as you have ring back)
 On Jul 24, 2013 9:20 PM, "Alex Mendoza"  wrote:

> Must check your routes
>
>
> Try pinging the ip phone's address from  CUPC PC.
>
> If it is unsuccessful do a tracert, to see which hop do not know how to
> reach the voice vlan.
>
>
> I think is easy to figure out what is going on.
>
> Best regards
>
> Alejandro Mendoza
> Sent from my iPhone 
>
> On 24/07/2013, at 20:12, Dharambir kumar varma 
> wrote:
>
> > Hi Team.
> >
> > i have one phone CUPC over internet...and one cisco 7941 phone internal..
> > both registered to call manager.
> >
> > when i call from cupc to 7941 or viceversa,,ring out happens and when
> > call is connected, only dead air/ No audio..
> > where can i check...
> > ___
> > 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
>
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Re: [OSL | CCIE_Voice] AAR and uccx

[Pavan K]

Ram,
If i remember correctly, the catch is that the AAR mask on the CTI port has
to point to the RP. so your aar mask on CTI port needs to be 4000 not 400x
On Jul 10, 2013 5:39 PM, "Ramcharan Arya"  wrote:

> Hi Piyush,
>
> I assigned HQ device pool and location setting on CTI route point and CTI
> ports to HQ.
>
> Following is isdn debug output when calling from 4002 to 4000 ( during AAR)
>
> HQ(config)#
> Jul 10 22:15:08.149: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8  callref =
> 0x000F
> Bearer Capability i = 0x8090A2
> Standard = CCITT
> Transfer Capability = Speech
> Transfer Mode = Circuit
> Transfer Rate = 64 kbit/s
> Channel ID i = 0xA98397
> Exclusive, Channel 23
> Calling Party Number i = 0x1181, '85224044011'
> Plan:ISDN, Type:International
> Called Party Number i = 0x91, '85224044102'
> Plan:ISDN, Type:International
> Jul 10 22:15:08.201: ISDN Se0/0/0:23 Q931: RX <- CALL_PROC pd = 8  callref
> = 0x800F
> Channel ID i = 0xA98397
> Exclusive, Channel 23
> HQ(config)#
> Jul 10 22:15:08.329: ISDN Se0/0/0:23 Q931: RX <- ALERTING pd = 8  callref
> = 0x800F
> Progress Ind i = 0x8088 - In-band info or appropriate now
> available
> Jul 10 22:15:08.341: ISDN Se0/0/0:23 Q931: TX -> DISCONNECT pd = 8
> callref = 0x000F
> Cause i = 0x8290 - Normal call clearing
>
> Site C phone 2 Line 2 4102 ring and call disconnect in 2 sec.
>
> with below error message
>
> Jul 10 22:16:48.524: ISDN Se0/0/0:15 Q931: TX -> SETUP pd = 8  callref =
> 0x0048
> Sending Complete
> Bearer Capability i = 0x8090A2
> Standard = CCITT
> Transfer Capability = Speech
> Transfer Mode = Circuit
> Transfer Rate = 64 kbit/s
> Channel ID i = 0xA9838B
> Exclusive, Channel 11
> Display i = 'SiteC Phone 2'
> Calling Party Number i = 0x1183, '+85224044002'
> Plan:ISDN, Type:International
> Called Party Number i = 0x91, '85224044010'
> Plan:ISDN, Type:International
> Jul 10 22:16:48.524: ISDN Se0/0/0:15 Q931: TX -> RELEASE pd = 8  callref =
> 0x80D5
> Jul 10 22:16:48.780: ISDN Se0/0/0:15 Q931: RX <- STATUS pd = 8  callref =
> 0x00D5
> Cause i = 0x80E202 - Message not compatible with call state or not
> implemented
> Call State i = 0x0C
> Jul 10 22:16:48.780: ISDN Se0/0/0:15 Q931: RX <- RELEASE pd = 8  callref =
> 0x8047
> Jul 10 22:16:48.780: ISDN Se0/0/0:15 Q931: RX <- RELEASE_COMP pd = 8
> callref = 0x00D5
>
> Basically when calling from HQ PH1 to 4000 ( in AAR) it work without any
> issue.
>
>
> Thanks,
> Ramcharan Arya
>
>
>
> On Tue, Jul 9, 2013 at 10:37 PM,  wrote:
>
>> Hello Ram,
>>
>> You can assign Hq device pool and location setting to cti route point and
>> cti ports..
>> And assign site c device pool and location to site c phones...
>>
>> Regards,
>> Piyush Jain
>>
>> Sent from my android device.
>>
>>
>>
>>
>> -Original Message-
>> From: Ramcharan Arya 
>> To: "ccie_voice@onlinestudylist.com" 
>> Sent: Tue, 09 Jul 2013 7:59 AM
>> Subject: [OSL | CCIE_Voice] AAR and uccx
>>
>> Hi,
>> I have a CTI Route point 4000 and two CTI port 410x and 410y
>>
>>
>>
>> SiteC Phone 1 and SC Phone 2 are in CSQ which is assign to application
>> and associated with trigger.
>>
>> Due to  RSVP when call exceed ip rsvp bandwidth call  from uccx to site
>> Phones should use AAR and to over PSTN.
>>
>> My doubts are .
>>
>> What should be local and device pool of CTI ports so it should work in
>> AAR when PSTN caller make call to CTI route point number 4000.
>>
>> Can someone please advice about this.
>>
>> Thanks,
>> Ramcharan Arya
>>
>>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
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>
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Re: [OSL | CCIE_Voice] Call Redirect - uccx

[Pavan K]

Check your redirect CSS on route point and CTI port
On Jul 4, 2013 2:27 PM, "Karen Johnson"  wrote:

>
> i script "Call redirect" to extension 5001 when press 1,  by hard code and
> variable, both of them did not work.
> Usually pretty simple and just work, but not this time.
>
> Each time I press 1 , just said "Please try again" .
>
> What is most likely  I miss here in configuration?
>
> Script:
> ---
> - Accept
> - Menu, press 1 , call redirect 5001
> - Select Resource CSQ
> - End
>
> tks
>
>
> ___
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> visit www.ipexpert.com
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Re: [OSL | CCIE_Voice] Lab Query

[Pavan K]

Yes.
On Jul 1, 2013 9:45 PM, "Kapuria, Aman" 
wrote:

> Hey Guys,
>
> ** **
>
> Do we have access to the Help page within the CUCM in lab? Do they block
> it? Can you click on service parameters to get the description?
>
> ** **
>
> *Aman Kapuria**  *
>
> ** **
>
> ___
> 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] Phone not pick up TFTP or config from UCM

[Pavan K]

Karen,
Check if your phone has an ITL/CTL installed. Erase them and see if the
changes take affect
Do you see the phone as registered in UCM  ?
On Jun 12, 2013 5:23 PM, "Randall Crumm"  wrote:

> Did you try to auto register the phones?
>
> Have a great day!
>
> Thanks,
> Randall
>
>   --
>  *From:* Karen Johnson 
> *To:* Bill Lake 
> *Cc:* "ccie_voice@onlinestudylist.com" 
> *Sent:* Wednesday, June 12, 2013 1:18 PM
> *Subject:* Re: [OSL | CCIE_Voice] Phone not pick up TFTP or config from
> UCM
>
> hi Bill,
>
> yes i got it from SC router, and i tried to ping UCM as well from the
> voice int vlan of SC and it works for the ping. but still not get the DN
>
> k
>
>   *From:* Bill Lake 
> *To:* Karen Johnson 
> *Cc:* "ccie_voice@onlinestudylist.com" 
> *Sent:* Wednesday, June 12, 2013 11:55:37 AM
> *Subject:* Re: [OSL | CCIE_Voice] Phone not pick up TFTP or config from
> UCM
>
>  Where are you getting your dhcp at sc?  If it is sc rtr then you could
> have it right and just no access to CUCM
>
> So try ping (CUCM ip) source vlan (voice #) and confirm the voice vlan has
> connectivity to CUCM
>
> Sent from my iPhone
>
> On Jun 12, 2013, at 11:29 AM, Karen Johnson 
> wrote:
>
>
> hi all,
>
> Phones are getting the ip address and option 150 from SC router DHCP.
> However the phones do not show DN or extension. (only blue image
> background)
>
> When I checked the Network setting in phone (ip,subnet, TFTP all showing
> correctly)
>
> - What is the cause phone not able to pick up the DN config from UCM (in
> UCM show the phones registered with DN) ?
> - I have also restart the TFTP and UCM.
> - here is my diagram :
>
> phone --- switch--- SC router (DHCP)
>
> tks for help
>
>
>
>  ___
> For more information regarding industry leading CCIE Lab training, please
> visit http://www.ipexpert.com/
>
> Are you a CCNP or CCIE and looking for a job? Check out
> http://www.platinumplacement.com/
>
>
>
>
> ___
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> visit www.ipexpert.com
>
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> www.PlatinumPlacement.com
>
>
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> visit www.ipexpert.com
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Re: [OSL | CCIE_Voice] rtp location start time

[Pavan K]

I second that. Excellent experience at RTP. Wasn't impressed with the
proctors at SJC.
On May 1, 2013 7:52 PM, "STEPHEN FREEBERG"  wrote:

> RTP exams begin at 7:15 am, You should arrive no later than 7:00 am.
>
> David Blair is the proctor and in my opinion is the best proctor I have
> encountered at Cisco.
>
> Steve
>
> On May 1, 2013, at 5:06 PM, Karen Johnson 
> wrote:
>
> hi,
>
> anyone can share what start time in RTP and proctors there?
>
> tks
> ___
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> 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
>
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> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 79, Issue 8

[Pavan K]

Ram,

If its based on 9.0 train of UCM i doubt v4.0 would come out before June 2013.
I would expect ISR G2 and MPLS to be included in the new version with
RT endpoints.

Since Cisco seems to be adding CCIE tracks left and right, I would
also like to see a CCIE in Contact Center Technologies added but thats
just being too wishful ;)

Again this is not based on any insider info and is pure unadulterated
speculation

-Pavan




On Tue, Sep 4, 2012 at 1:25 PM, Ramcharan Arya  wrote:
> Hi,
>
> I checked https://www.ipexpert.com/Cisco/CCIE/Voice/Bootcamps their
> website also advertising CCIE Voice version 4.0.
>
> When I click on one of the link its showing version 3.0.
>
>  My guess is version 4.0 is coming soon because ipexpert start
> updating their website.
>
>
> Regards,
> Ramcharan Arya
> CCIE # 28926 (R&S)
>
>
>
>
> On Mon, Sep 3, 2012 at 7:55 PM,   
> wrote:
>> Send CCIE_Voice mailing list submissions to
>> ccie_voice@onlinestudylist.com
>>
>> To subscribe or unsubscribe via the World Wide Web, visit
>> http://onlinestudylist.com/mailman/listinfo/ccie_voice
>> or, via email, send a message with subject or body 'help' to
>> ccie_voice-requ...@onlinestudylist.com
>>
>> You can reach the person managing the list at
>> ccie_voice-ow...@onlinestudylist.com
>>
>> When replying, please edit your Subject line so it is more specific
>> than "Re: Contents of CCIE_Voice digest..."
>>
>>
>> Today's Topics:
>>
>>1. Re: CCIE Blueprint version change (Dan Quinlan (daquinla))
>>2. Voicemail SRST (Justin Barksdale)
>>3. UCCX (Bill Lake)
>>4. Re: Voicemail SRST (Dan Quinlan (daquinla))
>>
>>
>> --
>>
>> Message: 1
>> Date: Tue, 4 Sep 2012 00:12:54 +
>> From: "Dan Quinlan (daquinla)" 
>> To: Bikramjit Singh 
>> Cc: "ccie_voice@onlinestudylist.com" 
>> Subject: Re: [OSL | CCIE_Voice] CCIE Blueprint version change
>> Message-ID: <7c6fde39-79f8-48b9-a096-caed1d166...@cisco.com>
>> Content-Type: text/plain; charset="us-ascii"
>>
>> I have no inside info on this, but given that it wasn't announced at Cisco 
>> LIVE! this year, I'd expect that the next revision, if it happens in the 
>> next 12 months, will be 2900 and 3900 series gateways with PVDM3's, 9.x 
>> trains of software, and will include video endpoints of some fashion (8900 
>> or 9900 series phones, perhaps the Jabber for Windows Client, maybe even a 
>> VCS-registered endpoint such as an EX unit or the Jabber Video client). As 
>> for timing, Cisco LIVE! next year seems logical to me.
>>
>> Again, this is all a guess and not based on any knowledge or fact.
>>
>> DQ
>> d...@cisco.com
>>
>> Sent from my iPhone
>>
>> On Sep 3, 2012, at 1:51 PM, "Bikramjit Singh"  wrote:
>>
>>> Hi Folks,
>>>
>>> Is there anyone who has a logical prediction on when is the blueprint about 
>>> to change?
>>> Also, what kind of changes are we expecting; both from software and 
>>> hardware (voice gateways, pvdm, etc..) perspective.
>>>
>>> Thanks!
>>> ___
>>> For more information regarding industry leading CCIE Lab training, please 
>>> visit www.ipexpert.com
>>>
>>> Are you a CCNP or CCIE and looking for a job? Check out 
>>> www.PlatinumPlacement.com
>>
>>
>> --
>>
>> Message: 2
>> Date: Mon, 3 Sep 2012 20:15:49 -0400
>> From: Justin Barksdale 
>> To: "ccie_voice@onlinestudylist.com" 
>> Cc: "ccie_voice@onlinestudylist.com" 
>> Subject: [OSL | CCIE_Voice] Voicemail SRST
>> Message-ID: 
>> Content-Type: text/plain;   charset=us-ascii
>>
>> Steve,
>>
>> Voicemail while in SRST can be accomplished using the redirecting number in 
>> order to reach the required mailbox.  Alternate extension are not required.
>>
>> Justin Barksdale
>> CCIE #29866
>>
>>
>> Sent from my iPhone 4.
>>
>> On Sep 3, 2012, at 8:00 PM, ccie_voice-requ...@onlinestudylist.com wrote:
>>
>>> Voicemail SRST
>>
>>
>> --
>>
>> Message: 3
>> Date: Mon, 3 Sep 2012 19:41:48 -0500
>> From: Bill Lake 
>> To: CCIE Study 
>> Subject: [OSL | CCIE_Voice] UCCX
>> Message-ID:
>> 
>> Content-Type: text/plain; charset="iso-8859-1"
>>
>> Trying to install UCCX on laptop to practice scripting but while I
>> installed Server 2003 and did the registry update, it says
>>
>> "Installation of Cisco Unified Contact Center Express cannot be performed
>> on the current version of MCS OS Service Release.  Please upgrade the OS
>> image version to 2003.1.5a or higher and try again"
>>
>> Does anyone know how to overcome this issue. I tried the following
>> [image: Inline image 1]
>>
>> Bill
>> -- next part --
>> An HTML attachment was scrubbed...
>> URL: 
>> 
>> -- next part --
>> A non-text attachment was scrubbed...
>> Name: image.png
>> Type: image/png
>> Size: 13715 bytes
>> Desc: not available
>> URL: 
>>
>> --
>>
>> Message: 4
>> Date: Tue

Re: [OSL | CCIE_Voice] Is it possible to prevent users from dialing VM Pilot

[Pavan K]

I am using a SIP based integration not the legacy SK so no hunt pilot.
Just a route pattern to the SIP Trunk

-Pavan

On Tue, Sep 4, 2012 at 10:24 AM, William Affeldt
 wrote:
> Check the hunt pilot. What partition is it in?
>
> Sent from my iPhone
>
> On Sep 4, 2012, at 8:03 AM, Pavan K  wrote:
>
>> UCM 8.6.2 / Unity Connection 8.6.2
>>
>> Is there a way to prevent users from dialing the Voicemail pilot directly ?
>>
>>
>> I tried the following but it doesn't work
>> (1) Place the Unity Route Pattern in a Partition (lets say Voicemail)
>> (2) Configure the CSS on the Voicemail Pilot to include the Voicemail 
>> Partition
>> (3) Device/Line CSS on the Phone does not have the Voicemail pilot in it
>>
>> It seems to always use the Line/Device CSS to call into the Voicemail
>> System when
>> the Voicemail button is pressed on the phone (instead of using the CSS
>> configured in Voicemail Pilot / Profile assigned to the device).
>>
>>
>> Forwarded calls get routed to Voicemail as the CF CSS is set to use
>> "With Configured CSS" and the CSS for Forwarding has the Voicemail CSS
>>
>> --
>> - Pavan
>> ___
>> 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



-- 
- Pavan
___
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[OSL | CCIE_Voice] Is it possible to prevent users from dialing VM Pilot

[Pavan K]

UCM 8.6.2 / Unity Connection 8.6.2

Is there a way to prevent users from dialing the Voicemail pilot directly ?


I tried the following but it doesn't work
 (1) Place the Unity Route Pattern in a Partition (lets say Voicemail)
 (2) Configure the CSS on the Voicemail Pilot to include the Voicemail Partition
 (3) Device/Line CSS on the Phone does not have the Voicemail pilot in it

It seems to always use the Line/Device CSS to call into the Voicemail
System when
the Voicemail button is pressed on the phone (instead of using the CSS
configured in Voicemail Pilot / Profile assigned to the device).


Forwarded calls get routed to Voicemail as the CF CSS is set to use
"With Configured CSS" and the CSS for Forwarding has the Voicemail CSS

-- 
- Pavan
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Re: [OSL | CCIE_Voice] [cisco-voip] UCCX 9 Application User Password recovery ?

[Pavan K]

Thank you guys.

I was able to recover using Ryan's suggestion after resetting the password.

-Pavan



On Mon, Aug 27, 2012 at 7:41 AM, Ryan LaFountain (rlafount)
 wrote:
> Hi Pavan,
>
> You can use the Application User that was set during install to get into
> the GUI at any time in UCCX 9.0(1). If you have forgotten that password,
> you can reset it using the CLI commands found in the CLI Guide (the same
> as if you forgot the Application User in CCM).
>
> This user doesn't require authentication to CUCM, so you can use it to
> reset the AXL Provider user and designate more CCM End Users as UCCX
> Admins.
>
> HTH.
>
> Thank you,
>
> Ryan LaFountain
> Unified Contact Center
> Cisco Services
> Direct: +1 919 392 9898
> Email: rlafo...@cisco.com
> Hours: M ­ F 9:00am ­ 5:00pm
>
>
>
>
> On 8/26/12 11:07 PM, "Pavan K"  wrote:
>
>>I have a test UCCX 9 server that was configured and operational.
>>The UCM cluster that was integrated with the UCCX was re-installed from
>>scratch.
>>
>>Is there a way to recover the application user password / switch the
>>server to the default post install state
>>Basically trying to figure out a way to get into the appadmin gui.
>>
>>--
>>- Pavan
>>___
>>cisco-voip mailing list
>>cisco-v...@puck.nether.net
>>https://puck.nether.net/mailman/listinfo/cisco-voip
>



-- 
- Pavan
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[OSL | CCIE_Voice] UCCX 9 Application User Password recovery ?

[Pavan K]

I have a test UCCX 9 server that was configured and operational.
The UCM cluster that was integrated with the UCCX was re-installed from scratch.

Is there a way to recover the application user password / switch the
server to the default post install state
Basically trying to figure out a way to get into the appadmin gui.

-- 
- Pavan
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[OSL | CCIE_Voice] CME SIP Phone -- Unity Connection DTMF issue

[Pavan K]

CME SIP Phone is in remote site with cRTP enabled between the CME & HQ
sites so RFC2833 / rtp-nte is un-usable.

The only other option on CME SIP Phones is sip-notify.
The available options on the dial-peer are sip-notify & sip-kpml.
The only option on Unity Connection is sip-kpml (RFC2833 cannot be used).

Everything i have read / tried seems to indicate that sip-notify /
sip-kpml do not work (even if using an XCODER).

http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/integration/cucme_sip/guide/cucintcucmesip030.html#wp1094879


Did anybody get sip-kpml / sip-notify to work with Unity connection  from CME ?

-- 
- Pavan
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[OSL | CCIE_Voice] Can CME SK phones be forced to g711 only ?

[Pavan K]

If an ephone has codec g711ulaw hard-coded on it and the inbound call
comes in as a g729 call,
call is getting connected as  g729r8 without transcoder.

Is there a way to disable this behavior ?

codec g711ulaw does not allow dspfarm-assist to be configured


ephone-1[0] Mac:000F.23FC.A637 TCP socket:[2] activeLine:0 REGISTERED
in SCCP ver 12/9
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0
debug:0 caps:8 privacy:1
IP:142.103.66.52 51008 7970  keepalive 167 max_line 8
button 1: dn 1  number 3001 CH1   IDLE CH2   IDLE CH3
 IDLE CH4   IDLE CH5   IDLE CH6   IDLE
CH7   IDLE CH8   IDLE
Preferred Codec: g711ulaw
Username: sk1 Password: cisco


-- 
- Pavan
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[OSL | CCIE_Voice] [Resolved] UCCX - Do CTI Ports in Partition work ?

[Pavan K]

Thanks Gurpreet, Dan & Krishna.

This is now fixed. As Dan mentioned the CSS of caller matters.

==
When a CTI Route point, redirects the call to the CTI port, the CSS of
the device that calls the Route point is used to search for the CTI
Port.
==



From: Pavan K 
Date: Mon, Jun 11, 2012 at 1:33 PM
To: ccie_voice@onlinestudylist.com


With UCCX, did anybody get calls to work when the CTI ports are in a
partition ?
If so what CSS did you have to configure ?

I have created a RoutePoint in the NULL Partition and CTI ports in a UCCX
partition
Added a CSS for the RoutePoint that includes the UCCX partition (on both the
line & device) but the call doesn't connect.

If i take the CTI ports out of the partition, everything works perfectly.

TIA
--
- Pavan

--
From: Gurpreet Singh Kukreja 
Date: Mon, Jun 11, 2012 at 3:27 PM
To: Pavan K 
Cc: ccie_voice@onlinestudylist.com


Hi Pavan,

We've seen this behavior with UCCX.

Logically, the calls should work w/ or w/o partition applied on the CTI Port
Group, keeping in mind the CSS applied on the CTI Route Point.

Few things to keep in mind:

1) Always apply the changes on these Triggers/ Port Groups from the CCX and
never from the CM.
2) If you apply the correct CSS on the Trigger which includes the partition
of the Port group, the calls should work.
3) Even after applying the changes if the calls do not work, it could be
very possible that the changes you're making from the CCX are not getting
updated on the CM. In this case, first run the Data Resync from the CCX and
make sure there are no exceptions in the output. Then, restart the CTI
Manager on all CM servers and then restart the CCX Engine.


- Gurpreet
>
> ___
> 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



--
From: Dan Quinlan (daquinla) 
Date: Mon, Jun 11, 2012 at 3:34 PM
To: Pavan K 
Cc: ccie_voice@onlinestudylist.com


I would think that the inbound caller (ip phone or gw) would need the CSS to
access the CTI ports.

DQ
d...@cisco.com

Sent from my iPhone

--
From: Krishna 
Date: Mon, Jun 11, 2012 at 3:53 PM
To: Pavan K 


pavan,

I worked on uccx lab and it worked fine for me. All that you need to
remember one point always, what does the CTI Route point has to see. in this
case the CTI route point has to see the phones partition in order to
handover the calls to the phone agents. Check that internal dns are listed
in your css to make this work.

thank you
krishna.

____
From: Pavan K 
To: ccie_voice@onlinestudylist.com
Sent: Monday, June 11, 2012 1:33 PM
Subject: [OSL | CCIE_Voice] UCCX - Do CTI Ports in Partition work ?
___
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--
- Pavan
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Re: [OSL | CCIE_Voice] no trace found but calls get routed br1<->Hq

[Pavan K]

Krishna,

When dialing from station to station and both stations are registered to
UCM,
the call does not normally traverse through the PSTN (no AAR case).
The signaling & media flows over voip directly which is why you dont see
any gateway / q931 debugs being active.

 However for a Voip flow to maintain proper quality, CAC/RSVP is used to
ensure sufficient bandwidth being used which is why you see the RSVP debug
active.

Media flows from endpoint to endpoint directly through the RSVP agents
which is what you see in "sh sccp connections"

Signaling flows from endpoint to UCM direct. Remember the gateway is not in
the signaling path which is why you do not see anything on the gw.



On Mon, Jun 11, 2012 at 11:42 AM, Krishna  wrote:

> Hi folks,
>
> I couldn't understand the call flow between HQ and BR1 which are
> provisioned/registed in the cucm. here is the detail structure:
>
> HQ-phone1 -5002
> css-hq-international
> pt-pt-internal
>
> BR1-phone1-1002
> css-br1-ld
> pt-pt-internal
>
> Both phones are residing in the partition pt-internal, and br1 is a mgcp
> site and whereas the hq is the h323 site. when i call 1002 from 5002 or
> vice versa the call works fine, but when i enable deb isdn q931 or deb voip
> dialp, i dont see anything. Whereas when i enable RSVP based CAC, i can see
> the traces with the show sccp connections.
>
> could any one help me out how the calls are working in between these two.
> is it because the phones are registered to cucm, but logically in a
> different device pool and therefore it routes directly on cucm your
> help is much appreciated.
>
> Thank you.
>
> Krishna.
>
> ___
> 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
>



-- 
- Pavan
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[OSL | CCIE_Voice] UCCX - Do CTI Ports in Partition work ?

[Pavan K]

With UCCX, did anybody get calls to work when the CTI ports are in a
partition ?
If so what CSS did you have to configure ?

I have created a RoutePoint in the NULL Partition and CTI ports in a UCCX
partition
Added a CSS for the RoutePoint that includes the UCCX partition (on both
the line & device) but the call doesn't connect.

If i take the CTI ports out of the partition, everything works perfectly.

TIA
-- 
- Pavan
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[OSL | CCIE_Voice] Directed Call Park Range + BLF Question

[Pavan K]

Ok,

If i create a DPark number (say 5100) then i have no problem assigning it
to a BLF DPark Softkey.

However if i create a range of DPark numbers (say 510[0-3] or 510[0123] )
then the numbers do not show up in the drop down while assigning them to a
BLF DPark Softkey. Also there is no box to enter a freeform number.

Any trick to get this to work ?


-- 
- Pavan
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[OSL | CCIE_Voice] Custom CData on IP Phone Agent

[Pavan K]

If asked to add custom fields under the "Caller Data Softkey" in IP Phone
Agent, how does one do that ?

This is easily doable in CAD using the CDA & "Set Enterprise Call Info
command"

How does one do it on the IPPA ?
-- 
- Pavan
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Re: [OSL | CCIE_Voice] Changing NTP Timezone on UCM Pub for OS Admin ?

[Pavan K]

Thanks Baktha. That did it.

-Pavan

On Tue, Apr 10, 2012 at 10:56 AM, Baktha Muralidharan
wrote:

>  show timezone list
>  set timezone
>
> but I don't think you need to worry about, as far as times displayed on
> the phones. DTGs will take care of that.
>
> thanks,
> /Baktha
>
>
> Message: 3
> Date: Tue, 10 Apr 2012 09:39:42 -0500
> From: Pavan K 
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] Changing NTP Timezone on UCM Pub for OS
>Admin ?
> Message-ID:
><
> CAJDPBuVfgGzp5HNtYMrcUXfZYJ53jZU7negpE2q--ztE=y0...@mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> When looking at  "utils ntp status" on UCM pub, i see the Timezone as CST.
> Is there a way to change the Timezone from Amer/Chicago to a diff Zone.
>
> ___
> 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
>



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

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

[OSL | CCIE_Voice] delayed Multicast MOH streaming - Any ideas ?

[Pavan K]

Multicast MOH from CCM. No Spoofing.

For HQ site, MMOH stream perfectly
For Branch sites, on the Branch router i can see MMOH packets coming in with
the "debug ip mpacket" command but the phone doesnt play MOH until about 5
mins later.

In other words, after pressing Hold, Phone starts streaming music after
being on hold for 5 mins.

Phones are SK registered to CCM. CCM counters look good (show mcast stream
active)

pim is configured for sparse-dense-mode on all serial subifs, vlans &
loopbacks.

IOS is 12.4 (20T2)


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

Re: [OSL | CCIE_Voice] Missed call redial + dialing

[Pavan K]

Q1 if you want to try it out, it should display as 4 digits but call list
should have full e164.
Q2 You can put a TP in HQ phone 1 CSS to exapnd calling number to e164.

But really there are tons of possibilities & ways of accomplishing this
(gateway translations, transformations)
depends on what you want.

On Tue, Nov 9, 2010 at 12:19 PM, Shrini  wrote:

>  Hi Mike,
> Ans_to_Q1 : I am not following any specific question, just testing and
> wanted to understand how it should display.
> Ans_to_Q2 : If I manipulate at gateway level all other calls are affected
> and all calls ANI will be displayed as +14082011001 including where I want
> ANI to be 2011001, so setting at GW does not work.
>
> -Shrini
>
> On 11/9/2010 10:10 AM, Mike Nipp (mnipp) wrote:
>
>  Shini,
>
>
>
> Q: When I redial a missed call +19193012001 (on HQ phone 1001) Br1 Phone
> 2001 is ringing and is displaying number as from 1001. Is this as per lab
> requirement or does it need to display +14082011001 ?
>
> A: What is the question you to display?
>
>
>
> Q: Since redial of missed call matches route pattern \+! directly goes to
> GW. Where should I set so that calling party should displayed as
> +14082011001.
>
> A: You can apply a calling party transformation at the egress gateway to
> manipulate the calling number to the PSTN. This will change 1001 to
> +14082011001.
>
>
>
>
>
>
>
> *From:* ccie_voice-boun...@onlinestudylist.com [
> mailto:ccie_voice-boun...@onlinestudylist.com]
> *On Behalf Of *Shrini
> *Sent:* Sunday, November 07, 2010 7:08 PM
> *To:* ccie_voice@onlinestudylist.com
> *Subject:* [OSL | CCIE_Voice] Missed call redial + dialing
>
>
>
> Greetings Experts:
>
> When I redial a missed call +19193012001 (on HQ phone 1001) Br1 Phone 2001
> is ringing and is displaying number as from 1001. Is this as per lab
> requirement or does it need to display +14082011001 ?
>
> Since redial of missed call matches route pattern \+! directly goes to GW.
> Where should I set so that calling party should displayed as +14082011001.
>
> TIA
> Shini
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>


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

[OSL | CCIE_Voice] Unity connection Broadcast message MWI

[Pavan K]

Is there a way to get MWI for broadcast messages (sent using Broadcast
admin)
The message is in the mailbox but no MWI for these messages.

MWI works normally in other cases.

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

Re: [OSL | CCIE_Voice] [Q] SRST & DHCP in real world.

[Pavan K]

Agreed.
My question was can it be done if DHCP for Branch1 was scoped from HQ.
Looks like it cannot.

-Pavan

On Fri, Nov 5, 2010 at 6:07 PM, Shrini  wrote:

>  IOS DHCP on Br1 router is required.
>
> You will provide option 150 as CUCM IP address under DHCP pool.
> Also you are configuring SRST ref on CUCM which means Callmanger tell the
> devices if I fail get the config from router.
>
> Hope this clarifies.
>
>
>
> On 11/5/2010 7:24 AM, Pavan K wrote:
>
> Lets say we are required to configure Branch1 to be an SRST site.
> Also during normal operation, Branch1 phones get DHCP from CCM in HQ.
>
>  Now when BR1 is in SRST,
>
>  If any of the phones reset, they will not be able to get DHCP and will be
> unable to register to SRST.
> Does this question imply an additional DHCP configuration in BR1 site ?
> If so, is there any way to make the HQ DHCP primary and BR1 DHCP as
> fallback when the HQ DHCP pool is in-accessible.
>
>
>  How do people deal with this in real world (other than having local DHCP
> on BR1) ?
>
> --
> - Pavan
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please 
> visit www.ipexpert.com
>
>


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

Re: [OSL | CCIE_Voice] [Q] SRST & DHCP in real world.

[Pavan K]

Thanks for all the input folks.
Yes i did notice that reseting a phone from CLI causes the phone to lose its
IP.

-Pavan


On Fri, Nov 5, 2010 at 5:31 PM, George Goglidze  wrote:

> nobody will keep an IP if there is no DHCP present.
> Neither Phone, nor PC (Win/Linux/Mac).
>
> Regards,
>
>
>
> On Fri, Nov 5, 2010 at 3:13 PM,  wrote:
>
>> If I am not mistaken they would keep their previous address, the only it
>> would try to get a new ip if it was a new phone or there was a factory reset
>> or the network settings were changed, other than that they function normally
>> so it is not an issue.
>> -Original Message-
>> From: Pavan K 
>> Sender: ccie_voice-boun...@onlinestudylist.com
>> Date: Fri, 5 Nov 2010 09:24:57
>> To: osl osl
>> Subject: [OSL | CCIE_Voice] [Q] SRST & DHCP in real world.
>>
>> ___
>> 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
>>
>
>


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

[OSL | CCIE_Voice] [Q] SRST & DHCP in real world.

[Pavan K]

Lets say we are required to configure Branch1 to be an SRST site.
Also during normal operation, Branch1 phones get DHCP from CCM in HQ.

Now when BR1 is in SRST,

If any of the phones reset, they will not be able to get DHCP and will be
unable to register to SRST.
Does this question imply an additional DHCP configuration in BR1 site ?
If so, is there any way to make the HQ DHCP primary and BR1 DHCP as fallback
when the HQ DHCP pool is in-accessible.


How do people deal with this in real world (other than having local DHCP on
BR1) ?

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

Re: [OSL | CCIE_Voice] GateKeeper Problems

[Pavan K]

Ok, i missed the traces you attached at the very bottom. From the below line
it is very clear that you either dont have a CSS assigned or did not reset
after assigning CSS as the pss field is blank.

and as warren pointed out the most important keyword one is looking for is
the "BLOCK" and not the "NoPotentialMatchesExist"

Digit Analysis: getDaRes -
voiceMailCallingSearchSpace=[]|
08/28/2010 08:50:01.090 CCM|Digit analysis: match(pi="2",fqcn="",
cn="3003", plv="5", *pss=""*, TodFilteredPss="",
*dd="1#5004"*,dac="0")|

Regards

-Pavan


On Sat, Aug 28, 2010 at 4:10 PM, Warren Heaviside (wheavisi) <
wheav...@cisco.com> wrote:

> When interpreting SDI traces and you see "NoPotentialMatchesExist" it's
> a bit misleading.  What it actually means is "no more" potential matches
> exist and the Digit Analysis process is complete and has made a match.
>
> Warren
>
> Warren Heavisidewheav...@cisco.com
> ENGINEER.CUSTOMER SUPPORT
> High Touch Technical Support
> Phone: +1 408 853 7995
> Office Hour 9 am - 5 pm Pacific Monday - Friday
>
> For corporate legal information go to:
> http://www.cisco.com/web/about/doing_business/legal/cri/index.html
>
>
> -Original Message-
> From: ccie_voice-boun...@onlinestudylist.com
> [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of
> ccie_voice-requ...@onlinestudylist.com
> Sent: Saturday, August 28, 2010 1:41 PM
> To: ccie_voice@onlinestudylist.com
> Subject: CCIE_Voice Digest, Vol 54, Issue 95
>
> Send CCIE_Voice mailing list submissions to
>ccie_voice@onlinestudylist.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
>http://onlinestudylist.com/mailman/listinfo/ccie_voice
> or, via email, send a message with subject or body 'help' to
>ccie_voice-requ...@onlinestudylist.com
>
> You can reach the person managing the list at
>ccie_voice-ow...@onlinestudylist.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of CCIE_Voice digest..."
>
>
> Today's Topics:
>
>   1. Re: GateKeeper Problems (ccieid1ot)
>   2. Re: GateKeeper Problems (Pavan)
>   3. Proctorlabs.com is not loading (Aug 28,   1600 hours EST)
>  (David Lee)
>   4. Re: Proctorlabs.com is not loading (Aug 28,   1600 hours EST)
>  (David Lee)
>   5. proctor lab down ??? (Erwan Erwan)
>
>
> --
>
> Message: 1
> Date: Sat, 28 Aug 2010 13:12:15 -0500
> From: ccieid1ot 
> To: Josmar Ramirez 
> Cc: ccie_voice@onlinestudylist.com,
> ccie_voice-boun...@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] GateKeeper Problems
> Message-ID:
>
> 
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
> Yes, digit manipulation or sig dig 4.
>
> On Fri, Aug 27, 2010 at 9:42 PM, Josmar Ramirez
> wrote:
>
> > Of course I meant on the incoming digits on the ccm trunck
> >
> > - Original Message -
> > From: Josmar Ramirez
> > To: 'edot...@ams.net' ; 'pav.c...@gmail.com' <
> > pav.c...@gmail.com>; 'ccie_voice-boun...@onlinestudylist.com' <
> > ccie_voice-boun...@onlinestudylist.com>; 'dcrayto...@comcast.net' <
> > dcrayto...@comcast.net>
> > Cc: 'ccie_voice@onlinestudylist.com' 
> >  Sent: Fri Aug 27 22:39:20 2010
> > Subject: Re: [OSL | CCIE_Voice] GateKeeper Problems
> >
> > On the callmanager make sure you set the significant digits to 4 so
> that it
> > strips the 1# when it hits the trunk on ccm. Careful this might break
> any
> > teho config that you might be sending to ccm.
> >
> >
> > - Original Message -
> > From: ccie_voice-boun...@onlinestudylist.com <
> > ccie_voice-boun...@onlinestudylist.com>
> > To: Pavan ; ccie_voice-boun...@onlinestudylist.com
> <
> > ccie_voice-boun...@onlinestudylist.com>; dcrayto...@comcast.net <
> > dcrayto...@comcast.net>
> > Cc: ccie_voice@onlinestudylist.com 
> > Sent: Fri Aug 27 21:55:40 2010
> > Subject: Re: [OSL | CCIE_Voice] GateKeeper Problems
> >
> > Are you doing any digit translation to remove the prefix. That was
> tripping
> > me up.
> > Sent from my Verizon Wireless BlackBerry
> >
> > -Original Message-
> > From: Pavan 
> > Sender: 
> > Date: Fri, 27 Aug 2010 20:19:04
> > To: dcrayto...@comcast.net
> > Cc: ccie_voice@onlinestudylist.com
> > Subject: Re: [OSL | CCIE_Voice] GateKeeper Problems
> >
> > ___
> > 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
> >
> -- next part --
> An HTML attachment was scrubbed...
> URL:
>  >
>
> --
>
> Message: 2
> Date: Sat, 28 Aug 2010 13:25:46 -0500
> From: Pa

[OSL | CCIE_Voice] 3750 Sw - Queuing Threshold2 < Threshold1. What would happen ?

[Pavan K]

Folks,

When Configuring QOS Thresholds on the 3750 Switch,
I noticed that the switch allows us to configure Threshold 2 to be a lower
number than Threshold 1


What would happen in this scenario ?

Would COS/DSCP levels assigned to Threshold 2 be dropped at 10% while those
assigned to Threshold 1 be dropped at 20%
or
Something else ?


HQ-SW(config)#mls qos srr-queue input thres
HQ-SW(config)#mls qos srr-queue input threshold 1 20 10
HQ-SW(config)#do sh mls qos input-q
Queue :   1   2
--
buffers   :90  10
bandwidth :  90  10
priority  : 10   0
threshold1:  *20* 100
threshold2:  *10* 100


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

Re: [OSL | CCIE_Voice] NTP on UCCX

[Pavan K]

you generally use them if you are doing NTP authentication and using ACL to
control who your NTP peers are.
Doesn't hurt to add but technically they are not required.

I usually add them

-Pavan

On Fri, Jun 11, 2010 at 10:31 AM, Dani Bug  wrote:

> thx i forgot to add these cmd
> but wondering if we need to add
> ntp source loopback0
> ntp server x.x.x.x source int fa0/0.100
> these cmd also required ??
>
>
> On Fri, Jun 11, 2010 at 11:27 AM, Pavan K  wrote:
>
>> you would also need clock timezone and summertime (if asked)
>>
>>
>> On Fri, Jun 11, 2010 at 10:22 AM, Dani Bug wrote:
>>
>>> go with option 2 NTP UCCX
>>> also i just want to confirm in NTP on R1 R2 and R3 which commands we
>>> required lab perspective view ...
>>>
>>> ntp server x.x.x.x
>>> wht else command we required to configure 
>>>
>>> Thx
>>> Dani
>>>
>>>  On Fri, Jun 11, 2010 at 9:28 AM, Pavan K  wrote:
>>>
>>>>  When configuring NTP on the UCCX server, i see two approaches when
>>>> asked to configure NTP.
>>>>
>>>> Which way would you go ?
>>>>
>>>>
>>>> - Configure NTP on the Windows OS (Using windows registry hack) (More
>>>> Involved)
>>>>
>>>> - Configure NTP on the UCCX app itself during Integration. (Faster &
>>>> Seems to work)
>>>>(Disadvantage : windows time does not seem to sync up since only app
>>>> is synced)
>>>>
>>>>
>>>> --
>>>> - Pavan
>>>>
>>>> ___
>>>> For more information regarding industry leading CCIE Lab training,
>>>> please visit www.ipexpert.com
>>>>
>>>>
>>>
>>
>>
>> --
>> - Pavan
>>
>
>


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

[OSL | CCIE_Voice] CUPC and presence status

[Pavan K]

CUPC installed and working. It is not integrated into AD.

I can view status between two CUPC users (i.e status of user1 in CUPC2 and
vice versa
If i create my own contacts (Local contacts) on CUPC, should i be able to
view their presence status ?

Subscribe CSS on SIP trunk has been set appropriately.

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

Re: [OSL | CCIE_Voice] Better Voice Lab Locations

[Pavan K]

In SJ they generally take you to the cafe.
In RTP they generally get the food catered and serve it inside an ancillary
room. (atleast they used to)

On Fri, Jun 11, 2010 at 10:26 AM, Jeff Price (jeffpric)
wrote:

> I believe it depends on your location, but normally they walk you to a
> local Cisco cafeteria with a voucher for your lunch (up to a certain
> price).
>
> -Original Message-
> From: ccie_voice-boun...@onlinestudylist.com
> [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jon1992
> Sent: Friday, June 11, 2010 4:10 AM
> To: Amp; ccie voice
> Cc: ccie_voice@onlinestudylist.com; Mouhammad Nasser
> Subject: Re: [OSL | CCIE_Voice] Better Voice Lab Locations
>
> During lunch are we stuck in the lab area or can we go and buy?
>
> --
> From: "Amp" 
> Sent: Thursday, June 10, 2010 11:01 PM
> To: "ccie voice" 
> Cc: ; "Mouhammad Nasser"
> 
> Subject: Re: [OSL | CCIE_Voice] Better Voice Lab Locations
>
> > No not based on lunch. With the longer lunch time I will be able to
> have
> > some time to think about what I have completed, what I need to
> complete,
> > and if I need to change anything that I have done.
> >
> > Quoting ccie voice :
> >
> >> @Amp
> >>
> >> So you choose a lab location based on lunch?
> >>
> >> On Thu, Jun 10, 2010 at 1:14 PM, Amp  wrote:
> >>
> >>> I live here in the RTP area but have decided to take the lab in San
> >>> Jose.
> >>> Here are my reasons:
> >>>
> >>> 1. Later Start Time
> >>> 2. Longer Lunch
> >>> 3. Better Weather
> >>> 4. Just have a gut feeling about SJC
> >>>
> >>> Amp
> >>>
> >>>
> >>> Quoting Jeff Garvas :
> >>>
> >>>  I heard that the West coast facility starts later, so someone east
> of
> >>> that
>  location would gain the time zone benefits as well as the late
> start.
>  RTP
>  supposedly starts first thing in the morning bright and early.
> 
>  2010/6/9 Mouhammad Nasser 
> 
>    Hi,
> >
> > I think it is better to take one that is closest to one's
> timezone!
> > this
> > will eliminate the factor of travel sickness, and one may go to
> exam
> > awake
> > enough!
> >
> >
> >
> > Regards,
> >
> > --
> > 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
> >
> >
> >
> 
> >>>
> >>>
> >>> ___
> >>> 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
>



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

Re: [OSL | CCIE_Voice] NTP on UCCX

[Pavan K]

you would also need clock timezone and summertime (if asked)


On Fri, Jun 11, 2010 at 10:22 AM, Dani Bug  wrote:

> go with option 2 NTP UCCX
> also i just want to confirm in NTP on R1 R2 and R3 which commands we
> required lab perspective view ...
>
> ntp server x.x.x.x
> wht else command we required to configure 
>
> Thx
> Dani
>
> On Fri, Jun 11, 2010 at 9:28 AM, Pavan K  wrote:
>
>> When configuring NTP on the UCCX server, i see two approaches when asked
>> to configure NTP.
>>
>> Which way would you go ?
>>
>>
>> - Configure NTP on the Windows OS (Using windows registry hack) (More
>> Involved)
>>
>> - Configure NTP on the UCCX app itself during Integration. (Faster & Seems
>> to work)
>>(Disadvantage : windows time does not seem to sync up since only app is
>> synced)
>>
>>
>> --
>> - Pavan
>>
>> ___
>> For more information regarding industry leading CCIE Lab training, please
>> visit www.ipexpert.com
>>
>>
>


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

[OSL | CCIE_Voice] NTP on UCCX

[Pavan K]

When configuring NTP on the UCCX server, i see two approaches when asked to
configure NTP.

Which way would you go ?


- Configure NTP on the Windows OS (Using windows registry hack) (More
Involved)

- Configure NTP on the UCCX app itself during Integration. (Faster & Seems
to work)
   (Disadvantage : windows time does not seem to sync up since only app is
synced)


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

[OSL | CCIE_Voice] CUE integration with CCM problems

[Pavan K]

If i place all my CTI ports in the NULL partition everything works, If i put
them in PT-VM i get BUSY (CTI rejecting call).

Any additional CSS settings needed ?

=
Placed all my CTI Ports in PT-VM
Placed all my CTI Routepoints in Null partition.

CSS for Routepoints and Ports is CSS-VM { PT-VM + PT-Phones }
Doesn't work
==










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

[OSL | CCIE_Voice] How to send a secure message in Unity Connection ?

[Pavan K]

I can send messages as "Private" / "urgent".
How does one send a "secure" message ?

I haven't been able to find any useful docs on this yet !



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

Re: [OSL | CCIE_Voice] VPIM error on CUE (554 Bad Sender's System) [Solved]

[Pavan K]

Had to change domain name on Unity connection under SMTP settings and reboot
the box.
Restarting the Conversation Manager service (as instructed by the GUI)
didn't make any difference.

-Pavan

On Sat, Jun 5, 2010 at 7:41 PM, Pavan K  wrote:

> Trying VPIM
>
> Sending messages from CUE to UnityConnection works perfectly.
> Messages from UnityConnection to CUE get an error message and generate a
> NDR (non-delivery receipt)
>
> Looking through the SMTP traces, i see a 554 error. (Screenshot attached).
>
>
> Anybody seen this before ?
>
>
> --
> - Pavan
>



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

Re: [OSL | CCIE_Voice] CME 7.0 Presence caller-list is not working ...

[Pavan K]

As others have pointed out it works fine on 7.0.
Just do a create cnf-files and reset if your SK phones dont show presence in
call-list.

-Pavan


On Sat, Jun 5, 2010 at 5:41 AM, Angel Perez  wrote:

>  Hi:
>
> Sometimes you have to reload the gw to make presence works
>
> hth
>
> --
> Date: Sat, 5 Jun 2010 12:18:43 +0200
> From: findko...@gmail.com
> To: salman.shaik...@gmail.com
> CC: ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] CME 7.0 Presence caller-list is not working
> ...
>
>
> and maybe
>
> sip-ua
>presence enable
>
> will help?
>
> On Sat, Jun 5, 2010 at 12:16 PM, kobel  wrote:
>
> try "create cnf-files" & restart the phones.
>
>
>   On Sat, Jun 5, 2010 at 4:21 AM, Shadow of Voice <
> salman.shaik...@gmail.com> wrote:
>
>   Hi Guys
>
> I have issue when configure presence in CME I allow subscribe and allow
> watch globally still can't see caller list on missed call does any one know
> where i am wrong and why my CME presence caller-list is not working
> !
> presence
>  presence call-list
>  allow subscribe
> !
> ephone-dn  2  octo-line
>  number 4002 no-reg primary
>  description +6524044002
>  name SiteC-Ph2
>  allow watch
>  call-forward busy 4220
>  call-forward noan 4220 timeout 20
> !
> !
> ephone  1
>  device-security-mode none
>  mac-address 001A.A1C8.0H8F
>  ephone-template 1
>  blf-speed-dial 1 4002 label "SiteC-Ph2"
>  type 7961
>  button  1:1 3:3 4:5
> !
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
>
>
> --
> Hotmail: Powerful Free email with security by Microsoft. Get it 
> now.
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>


-- 
- Pavan
___
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www.ipexpert.com

Re: [OSL | CCIE_Voice] Attendant console link ?

[Pavan K]

Thanks guys for the response.
I got access to the file.

I am using this for the lab exam so cant use third party.

-pavan

On Tue, Jun 1, 2010 at 1:43 PM, johan_claes  wrote:

>  better to buy peter connect for reception,
> cheaper and better,
>
> johan claes
> ccie#5437
>
> - Original Message -
> *From:* kerboute kerboute 
> *To:* Pavan K 
> *Cc:* ccie_voice@onlinestudylist.com
> *Sent:* Tuesday, June 01, 2010 10:29 AM
> *Subject:* Re: [OSL | CCIE_Voice] Attendant console link ?
>
> attendant console is end of life for CUCM 7
> You need Cisco unified attendant console server
>
>
> On 05/31/2010 11:28 PM, Pavan K wrote:
>
> Does anybody have a link to / copy of the attendant console plugin ?
>
> --Thanks in advance.
> - Pavan
>
>
> ___
> 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
>
>


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

Re: [OSL | CCIE_Voice] MVA on SIPGW

[Pavan K]

When using SIPGW and trying to transfer a call,

The INVITE with the diversion header reaches CCM but is getting blocked in
there due to a Top level domain mismatch.

Wondering if anybody got it to work ?

-Pavan


On Mon, May 31, 2010 at 1:55 PM, Pavan K  wrote:

> Has any body tried this ?
>
> --
> - Pavan
>



-- 
- Pavan
___
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[OSL | CCIE_Voice] Attendant console link ?

[Pavan K]

Does anybody have a link to / copy of the attendant console plugin ?

--Thanks in advance.
- Pavan
___
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[OSL | CCIE_Voice] MVA on SIPGW

[Pavan K]

Has any body tried this ?

-- 
- Pavan
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[OSL | CCIE_Voice] E164 normalization on SIP Trunk for inbound calls

[Pavan K]

Folks,



For inbound calls, we can normally prefix/strip  digits on the H323  / MGCP
gateway page based on the calling number type (subscriber / national / ... )

When a call comes in through a SIP trunk, we lose the number type (due to
SIP limitations).

Does anybody have a good idea to normalize / re-classify the incoming call
(subscriber / national ) in this scenario ?

I am using CCM 7.0


-- 
- Pavan
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[OSL | CCIE_Voice] CME - SIP - COR - Should i get an immediate re-order ?

[Pavan K]

When a sip phone (phone A) calls phone B and COR blocks the call.
Is phone A supposed to get an immediate re-order tone like its Skinny
buddies?


>From what i see,

PhA   CME(SIP)
 -> INV with SDP
 < 183 with SDP(CME IP/Port)
 < 183 with SDP(CME IP/Port)
..repeats couple of times
Timeout and reorder on PhA.


Is this how its supposed to work ?



-- 
---
Pavan Katta
CCVP, CCDP
Development Engineer - Enterprise Voice

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[OSL | CCIE_Voice] SIP & SK phones show different time in CME

[Pavan K]

I am running 8.3.2.27 on the SK phones. IOS is 12.4(20)T2 and CME 7.0(0)
The SK phone seems to be showing incorrect time.
Has anybody seen this before ?

-Pavan


===
voice register global
 mode cme
 source-address 142.103.66.254 port 5060
 max-dn 10
 max-pool 2
 load 7961GE term61.default
 load 7961 term61.default
 authenticate register
 timezone 43
 time-format 24
 tftp-path flash:
 create profile sync 0007641738744237
 ntp-server 142.3.64.254 mode anycast
==
ntp server 142.3.64.254
telephony-service
 load 7970 term70.default
 load 7971 term71.default
 max-ephones 2
 max-dn 10 no-reg
 ip source-address 142.103.66.254 port 2000
 system message Your current Options
 time-zone 43
 time-format 24
 max-conferences 12 gain -6
 transfer-system full-consult
 create cnf-files version-stamp 7960 Feb 27 2010 22:28:52
===
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