Re: [cisco-voip] OT: movie continuity guy should get fired

[Abebe Amare]

what was the movie

On Wed, Nov 15, 2017 at 11:09 PM, Lelio Fulgenzi  wrote:

>
> Not only is this phone _not_ registered, but these don't do full duplex
> speakerphone!
>
>
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>
>
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Re: [cisco-voip] python project ideas

[Abebe Amare]

Thanks for your input guys.

I will spin up CUCM and expressway VMs and get working.

best regards,

Abebe

On Fri, Oct 13, 2017 at 12:31 PM, Heim, Dennis <dennis.h...@wwt.com> wrote:

> Expressway API’s should be fairly easy.
>
>
>
> *Dennis Heim | Emerging Technology Architect (Collaboration)*
>
> World Wide Technology, Inc. | +1 314-212-1814 <+1%20314-212-1814>
>
> [image: cid:image001.png@01D10DD2.7FC81F90]
> <https://twitter.com/CollabSensei>
>
> [image: cid:image002.png@01D10DD2.7FC81F90][image:
> cid:image003.png@01D10DD2.7FC81F90] <+13142121814>[image:
> cid:image004.png@01D10DD2.7FC81F90]
>
> "Worry less about who you might offend, and more about who you might
> inspire" -- Tim Allen
>
> “When you have unlimited time, its easy” – Captain Chesley Sullenberger
>
> “There is a fine line between Wrong and Visionary. Unfortunately, you have
> to be a visionary to see it." – Sheldon Cooper
>
> “The greatest danger for most of us is not that our aim is too high and we
> miss it, but that it is too low and we reach it.” -- Michelangelo Buonarroti
>
> “We should transform the way we work” – Rowan Trollope
>
> “If you’re not failing every now and again, it’s a sign you’re not doing
> anything very innovative” – Woody Allen
>
>
>
> *Click here to join me in my Collaboration Meeting Room
> <https://wwt.webex.com/meet/dennis.heim>*
>
>
>
> *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
> Of *Abebe Amare
> *Sent:* Friday, October 13, 2017 11:13 AM
> *To:* cisco voip <cisco-voip@puck.nether.net>
> *Subject:* [cisco-voip] python project ideas
>
>
>
> Hi,
>
>
>
> I am taking Python for network engineers course and have to do a project
> on it. The project must be using an API or a major framework to do
> something. It ideally should involve manipulating real-world data.
>
> I was looking to do my project focusing on Cisco collaboration using REST
> API. Can you guys suggest any project ideas?
>
>
>
> Thanks in advance
>
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[cisco-voip] python project ideas

[Abebe Amare]

Hi,

I am taking Python for network engineers course and have to do a project on
it. The project must be using an API or a major framework to do something.
It ideally should involve manipulating real-world data.
I was looking to do my project focusing on Cisco collaboration using REST
API. Can you guys suggest any project ideas?

Thanks in advance
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Re: [cisco-voip] CUCM 10.5(2) ASA Phone Proxy

[Abebe Amare]

Hi Brian,

Thanks for the invaluable support.

regards,

Abebe

On Thu, Jul 7, 2016 at 5:20 PM, Brian Meade <bmead...@vt.edu> wrote:

> Really all the phone is trusting is the locally stored CTL on the ASA with
> just the ASA cert in there.  Since you're not using TLS to CUCM (non-secure
> cluster), you don't really need any CUCM certs on the ASA.
>
> On Thu, Jul 7, 2016 at 5:14 AM, Abebe Amare <abu...@gmail.com> wrote:
>
>> Hi Brian,
>>
>> The cluster is in non-secure mode. From the ASA side, it looks like I
>> have to change only the CUCM address in the phone proxy configuration
>> without downloading the Certificates again. Is my assumption correct?
>>
>> regards,
>>
>> Abebe
>>
>> On Tue, Jul 5, 2016 at 10:55 PM, Erick Bergquist <erick...@gmail.com>
>> wrote:
>>
>>> Yea, I stumbled across the ASA guide mentioning it when I was trying
>>> to find something stating CUCM 8.6 and phone proxy wasn't supported.
>>>
>>> On Tue, Jul 5, 2016 at 12:17 PM, Brian Meade <bmead...@vt.edu> wrote:
>>> > I'm not so sure that was supposed to be added in there.  Phone proxy
>>> never
>>> > supported the security by default features of CUCM which is why it
>>> went End
>>> > of Support with 8.x along with Phone VPN being launched.  It looks
>>> like a
>>> > doc bug was made to add CUCM 8.0 support into the ASA config guide
>>> recently-
>>> > https://bst.cloudapps.cisco.com/bugsearch/bug/CSCto66376
>>> >
>>> > Security By Default features were never added to the ASA code that I
>>> know
>>> > of- https://bst.cloudapps.cisco.com/bugsearch/bug/CSCti62447
>>> >
>>> > On Tue, Jul 5, 2016 at 1:19 PM, Erick Bergquist <erick...@gmail.com>
>>> wrote:
>>> >>
>>> >> The ASA 9.x documentation has Call Manager 8.0.x listed in it's
>>> >> configuration guide for phone proxy. Just went through this recently
>>> >> working on a phone proxy issue.
>>> >>
>>> >>
>>> >> On Tue, Jul 5, 2016 at 10:58 AM, Brian Meade <bmead...@vt.edu> wrote:
>>> >> > Technically phone proxy isn't supported on 8.x either.  It ended
>>> support
>>> >> > after 7.x and Phone VPN replaced it in 8.x.  If you're just using
>>> >> > 7940/60s
>>> >> > and IP Communicator, it should work still though.
>>> >> >
>>> >> > Do you have a mixed mode CUCM cluster now or just doing non-secure
>>> >> > between
>>> >> > the ASA and CUCM?  You can check the Cluster Security Mode under
>>> >> > System->Enterprise Parameters.
>>> >> >
>>> >> > You really will want to use Phone VPN or MRA with Expressway
>>> instead of
>>> >> > Phone VPN though as it's not supported by TAC unless on CUCM 7.x.
>>> >> >
>>> >> > On Tue, Jul 5, 2016 at 5:05 AM, Abebe Amare <abu...@gmail.com>
>>> wrote:
>>> >> >>
>>> >> >> I am on the planning process to migrate CUCM 8.5 cluster to 10.5(2)
>>> >> >> using
>>> >> >> PCD simple migration to minimize any change. Since Phone Proxy is
>>> not
>>> >> >> supported on CUCM 10.x, I am thinking to keep the 8.5 cluster but
>>> >> >> change the
>>> >> >> IP address. My question is this:
>>> >> >>
>>> >> >> 1. Do I have to enroll the certificate from CUCM to ASA when I
>>> change
>>> >> >> the
>>> >> >> IP address of CUCM 8.5?
>>> >> >> 2. What are other alternative features to phone proxy?
>>> >> >>
>>> >> >> best regards,
>>> >> >>
>>> >> >> Abebe
>>> >> >>
>>> >> >> ___
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>>> >> >> cisco-voip@puck.nether.net
>>> >> >> https://puck.nether.net/mailman/listinfo/cisco-voip
>>> >> >>
>>> >> >
>>> >> >
>>> >> > ___
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>>> >> > https://puck.nether.net/mailman/listinfo/cisco-voip
>>> >> >
>>> >
>>> >
>>>
>>
>>
>
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Re: [cisco-voip] Jabber phone mode outbound calls issue

[Abebe Amare]

I have encountered a similar problem in that I can't make outgoing calls to
PSTN from jabber. The solution was to make the capability on the voice
gateway voice port to voice only.

On Tue, Nov 24, 2015 at 10:34 PM, Ryan Huff  wrote:

> It sounds like your solution may be changing or removing application dial
> rules. Application dial rules add or remove digits to dialed numbers from
> applications that use CCM as the call control server (e.g Unified
> Communications Manager IM & Presence) and do not function like traditional
> patterns. After an application dial rule applies its rule logic to a remote
> (called) number, that number must still match an egress-able pattern (or
> other type of onnet/offnet pattern/DN).
>
>
> In practice, I do not use application dial rules for IM & Presence unless
> I have a specific use case; I typically allow the dialed digits from the
> Jabber client to enter CCM's numplan untreated by application dial rules.
> That said, I would not remove any of your application dial rules without
> first understanding why they are there and the impact it may have on other
> applications if they were to be removed.
>
>
> The other thing to keep in mind about application dial rules is that Call
> Manager does not use the *best match *algorithm that it uses with other
> patterns, it applies the first dialing rule matched, which is determined by
> the application dial rule's priority.
>
>
> You can research more on the topic on page 175 of
> http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/10_0_1/ccmcfg/CUCM_BK_C95ABA82_00_admin-guide-100.pdf
>
>
> Hope this helps,
>
>
> = Ryan =
>
>
>
> Email: ryanthomash...@outlook.com
>
> Spark: ryanthomash...@outlook.com
>
> Twitter: @ryanthomashuff 
>
> LinkedIn: ryanthomashuff 
>
> Web ryanthomashuff.com
>
>
> --
> *From:* abbas wali 
> *Sent:* Tuesday, November 24, 2015 2:05 PM
> *To:* 'Ryan Huff'; cisco-voip@puck.nether.net
>
> *Subject:* RE: [cisco-voip] Jabber phone mode outbound calls issue
>
>
> Its set to Standard Analysis.
>
>
>
> Something else I have noticed.
>
> There are application dial rules defined. On top (with top priority )
> there was Default rule beginning with blank, 0 digit strip and append 8.
>
>
>
> Some of the traces I found all my dials were appended by 8.
>
>
>
> Okay I have now moved the default dial rule to the bottom and all the
> correct one are on top.
>
>
>
> Now I can dial internally across cluster which is good. But cant dial
> external
>
>
>
> If that’s the case and have to define full dial plan in the app dial rule
> that will become quiet messy.
>
>
>
>
>
>
>
> *From:* Ryan Huff [mailto:ryanh...@outlook.com]
> *Sent:* 24 November 2015 18:30
> *To:* abbas wali ; cisco-voip@puck.nether.net
> *Subject:* Re: [cisco-voip] Jabber phone mode outbound calls issue
>
>
>
> Also and if it isn't too late, before you pull the SDL trace on the test
> call, can you verify that the *Digit Complexity Analysis* is set to 
> *TranslationAndAlternatePatternAnalysis
> *under *Service Parameters->Cisco Call Manager*?
>
>
>
> = Ryan =
>
>
>
>
>
> Email: ryanthomash...@outlook.com
>
> Spark: ryanthomash...@outlook.com
>
> Twitter: @ryanthomashuff 
>
> LinkedIn: ryanthomashuff 
>
> Web ryanthomashuff.com
>
>
> --
>
> *From:* cisco-voip  on behalf of Ryan
> Huff 
> *Sent:* Tuesday, November 24, 2015 1:00 PM
> *To:* abbas wali; cisco-voip@puck.nether.net
> *Subject:* Re: [cisco-voip] Jabber phone mode outbound calls issue
>
>
>
> I would deffinatley look into getting the clock sync to a strata 3 on the
> pub and then restart ntp services.
>
>
>
> Can you do a test call on one of the Jabber clients and pull of the SDL
> traces for the call?
>
>
>
>
>
> Sent from my T-Mobile 4G LTE Device
>
>
>
>  Original message 
> From: abbas wali
> Date:11/24/2015 12:13 PM (GMT-05:00)
> To: 'Ryan Huff' ,cisco-voip@puck.nether.net
> Subject: RE: [cisco-voip] Jabber phone mode outbound calls issue
>
> Hi Ryan,
>
>
>
> Thanks for the detailed response.
>
>
>
> Yes the issue is with Jabber clients and not the IP phones.
>
>
>
> The line itself which is shared with many devices, can make calls on any
> other device but fails when made from Jabber.
>
>
>
> I ran all the below Utils and all came out without any significant alarms
>
> The NTP, though is at stratum 4. But again that’s for both the clusters
> and one of them can make calls with jabber.
>
>
>
>
>
> Have ran some traces as below
>
> These are multiple failed calls.
>
> Not sure why there are so many REFER messages !!
>
>
>
> Thank s
>
>
>
>
>
> *From:* Ryan Huff [mailto:ryanh...@outlook.com ]
> *Sent:* 24 November 2015 15:32
> *To:* 

[cisco-voip] switch version from CUCM 9.1(1) to 10.5(2)

[Abebe Amare]

Hi,

I upgraded CUCM over the weekend from version 9.1 to 10.5(2) and applied
the upgrade license on the PLM for version 10.x .
I noticed some EM profiles, end users, phones and BLF speed dials
disappeared after the version switch over. Fortunately those missing items
were not much so I added them manually. what might have caused the
configuration missing ?

thanks in advance

Abebe
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Re: [cisco-voip] switch version from CUCM 9.1(1) to 10.5(2)

[Abebe Amare]

That is exactly what happened,

thanks Ryan, for the in-depth explanation.

Abebe,



On Tue, Oct 6, 2015 at 1:38 PM, Ryan Huff <ryanh...@outlook.com> wrote:

> My suspicion (and this is just a guess really, without more detail); is
> that after the 10.5 update was applied, but before the switch version was
> completed, changes where made to the data in the 9.1 version.
>
> When staging an upgrade in the inactive partition, changes made to the
> database in the active partition are not replicated to the database in the
> inactive partition. When you complete the switch version, the database from
> the inactive partition (created when you initially installed the upgrade)
> becomes active and any data differences between the databases are lost.
>
> The same thing would happen if you had to revert the upgrade and
> switch-version back to 9.1. Any changes made in 10.5 would not replicate
> into 9.1.
>
> Thanks,
>
> Ryan
>
> Sent from my iPad
>
> > On Oct 6, 2015, at 3:20 AM, Abebe Amare <abu...@gmail.com> wrote:
> >
> > Hi,
> >
> > I upgraded CUCM over the weekend from version 9.1 to 10.5(2) and applied
> the upgrade license on the PLM for version 10.x .
> > I noticed some EM profiles, end users, phones and BLF speed dials
> disappeared after the version switch over. Fortunately those missing items
> were not much so I added them manually. what might have caused the
> configuration missing ?
> >
> > thanks in advance
> >
> > Abebe
> > ___
> > cisco-voip mailing list
> > cisco-voip@puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
>
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Re: [cisco-voip] TEHO using SIP trunk ringback issue

[Abebe Amare]

Hi Nate,

I want to thank you for your support on this issue. The problem got
resolved after enabling PRACK on both SIP trunks.

Abebe

On Fri, Jul 24, 2015 at 5:41 PM, Abebe Amare abu...@gmail.com wrote:

 Hi Nate,

 When you say incoming h323 dial-peer, are you referring to the voip h323
 dial-peer from voice gateway to the call manager? I tried to enable the
 progress_ind commands under the incoming pots dial-peer but it refused.

 regards,

 abebe

 On Fri, Jul 24, 2015 at 5:21 PM, NateCCIE natec...@gmail.com wrote:

 I think incoming h.323 dial-peer.



 http://www.cisco.com/c/en/us/support/docs/voice/h323/22983-ringback.html





 *From:* Abebe Amare [mailto:abu...@gmail.com]
 *Sent:* Friday, July 24, 2015 7:58 AM

 *To:* NateCCIE
 *Cc:* cisco voip
 *Subject:* Re: [cisco-voip] TEHO using SIP trunk ringback issue



 Hi Nate,



 Just to confirm, those commands should be applied on the outgoing pots
 dial-peer?







 On Fri, Jul 24, 2015 at 4:38 PM, NateCCIE natec...@gmail.com wrote:

 That is it.  Just do it on both sides.



 Do you have

 progress_ind alert enable 8

 progress_ind progress enable 8

 progress_ind connect enable 8



 on your h.323 dial-peers?



 *From:* Abebe Amare [mailto:abu...@gmail.com]
 *Sent:* Friday, July 24, 2015 7:18 AM
 *To:* NateCCIE
 *Cc:* cisco voip
 *Subject:* Re: [cisco-voip] TEHO using SIP trunk ringback issue



 Hi Nate,



 On the SIP profile I enabled as follows



 [image: Inline image 1]



 Is there any other place where I enable PRACK?







 On Fri, Jul 24, 2015 at 3:38 PM, NateCCIE natec...@gmail.com wrote:

 Is PRACK enabled on the sip trunk?



 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On
 Behalf Of *Abebe Amare
 *Sent:* Friday, July 24, 2015 1:54 AM
 *To:* cisco voip
 *Subject:* [cisco-voip] TEHO using SIP trunk ringback issue



 Hi,



 The call flow is like this:



 CUCM cluster 1 SIP TrunkCUCM cluster 2---H323Voice
 gatewayISDN PRI---PSTN



 The problem is when calling from cluster 1 to the PSTN, the CUCM in
 cluster 2 plays ring back tone whether the final destination is
 reachable or not, we don’t hear the PSTN announcements in case the number
 is wrong or unreachable or…etc. How to remedy this.



 thanks in advance









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[cisco-voip] TEHO using SIP trunk ringback issue

[Abebe Amare]

Hi,

The call flow is like this:

CUCM cluster 1 SIP TrunkCUCM cluster 2---H323Voice
gatewayISDN PRI---PSTN

The problem is when calling from cluster 1 to the PSTN, the CUCM in cluster
2 plays ring back tone whether the final destination is reachable or
not, we don’t hear the PSTN announcements in case the number is wrong or
unreachable or…etc. How to remedy this.

thanks in advance
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Re: [cisco-voip] TEHO using SIP trunk ringback issue

[Abebe Amare]

Hi Nate,

Just to confirm, those commands should be applied on the outgoing pots
dial-peer?



On Fri, Jul 24, 2015 at 4:38 PM, NateCCIE natec...@gmail.com wrote:

 That is it.  Just do it on both sides.



 Do you have

 progress_ind alert enable 8

 progress_ind progress enable 8

 progress_ind connect enable 8



 on your h.323 dial-peers?



 *From:* Abebe Amare [mailto:abu...@gmail.com]
 *Sent:* Friday, July 24, 2015 7:18 AM
 *To:* NateCCIE
 *Cc:* cisco voip
 *Subject:* Re: [cisco-voip] TEHO using SIP trunk ringback issue



 Hi Nate,



 On the SIP profile I enabled as follows



 [image: Inline image 1]



 Is there any other place where I enable PRACK?







 On Fri, Jul 24, 2015 at 3:38 PM, NateCCIE natec...@gmail.com wrote:

 Is PRACK enabled on the sip trunk?



 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
 Of *Abebe Amare
 *Sent:* Friday, July 24, 2015 1:54 AM
 *To:* cisco voip
 *Subject:* [cisco-voip] TEHO using SIP trunk ringback issue



 Hi,



 The call flow is like this:



 CUCM cluster 1 SIP TrunkCUCM cluster 2---H323Voice
 gatewayISDN PRI---PSTN



 The problem is when calling from cluster 1 to the PSTN, the CUCM in
 cluster 2 plays ring back tone whether the final destination is reachable
 or not, we don’t hear the PSTN announcements in case the number is wrong or
 unreachable or…etc. How to remedy this.



 thanks in advance





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Re: [cisco-voip] TEHO using SIP trunk ringback issue

[Abebe Amare]

Hi Nate,

On the SIP profile I enabled as follows

[image: Inline image 1]

Is there any other place where I enable PRACK?



On Fri, Jul 24, 2015 at 3:38 PM, NateCCIE natec...@gmail.com wrote:

 Is PRACK enabled on the sip trunk?



 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
 Of *Abebe Amare
 *Sent:* Friday, July 24, 2015 1:54 AM
 *To:* cisco voip
 *Subject:* [cisco-voip] TEHO using SIP trunk ringback issue



 Hi,



 The call flow is like this:



 CUCM cluster 1 SIP TrunkCUCM cluster 2---H323Voice
 gatewayISDN PRI---PSTN



 The problem is when calling from cluster 1 to the PSTN, the CUCM in
 cluster 2 plays ring back tone whether the final destination is reachable
 or not, we don’t hear the PSTN announcements in case the number is wrong or
 unreachable or…etc. How to remedy this.



 thanks in advance



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Re: [cisco-voip] INCOMING PSTN CALLS UNKNOWN

[Abebe Amare]

Hi,

If you want to change the calling number on outgoing calls, you can do it
on the external phone number mask under the DN for the phone or in the call
party transformation under the route pattern.

hope that helps

Abebe

On Fri, Oct 24, 2014 at 12:09 PM, Eteng Okoi eteng.o...@unicem.com.ng
wrote:

 DEAR ALL,

 PLEASE I NEED YOUR SUPPORT.

 INCOMING CALLS APPEAR AS UNKNOWN NUMBER ON MY CISCO PHONE WHOSE
 CONFIGURATION IS AN FXO PORT
  AND
  ALSO CALLING OUT THROUGH E1 CARD SHOWS MY LOCAL CISCO IP PHONE EXTENSION
 NUMBER INSTEAD OF THE PILOT NUMBER.


 PLEASE I NEED YOUR HELP.

 THANKS.
 --
 Eteng Okoi
 Network Engineer
 Cell: 234(703)4153530 Ext:2601
 E-mail: eteng.o...@unicem.com.ng
 Website: www.unicem.com.ng

 If you are not the intended recipient and have received this e-mail in
 error, please delete it immediately from your system and notify the sender
 by e-mail or telephone. Be notified that you are not to copy, distribute or
 disclose this e-mail or any of its contents to any other party and any such
 action may be unlawful.
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[cisco-voip] ISDN call takes long time before connecting

[Abebe Amare]

Hi,

I am having a hard time convincing the local telco that outgoing calls to
PSTN numbers take about 11 seconds before the voice router receives
CALL_PROC from their switch. They insist this is normal and want to dismiss
the case but the delay is noticeable. Incoming calls have no problem.
The PRI line comes using microwave media due to lack of copper last mile to
the site and it connects to Cisco 2901 voice router which connects to CUCM
using SIP.
I have sent them the output of 'debug isdn q931' as follows:

*Oct 11 11:48:12.999: ISDN Se0/2/0:15 Q931: Applying typeplan for sw-type
0x12 is 0x2 0x1, Calling num x
*Oct 11 11:48:12.999: ISDN Se0/2/0:15 Q931: Sending SETUP  callref = 0x00C2
callID = 0x8043 switch = primary-net5 interface = User
*Oct 11 11:48:13.003: ISDN Se0/2/0:15 Q931: TX - SETUP pd = 8  callref =
0x00C2
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA9839F
Exclusive, Channel 31
Calling Party Number i = 0x2180, 'x'
Plan:ISDN, Type:National
Called Party Number i = 0x81, 'y
Plan:ISDN, Type:Unknown
*Oct 11 11:48:13.295: ISDN Se0/2/0:15 Q931: RX - SETUP_ACK pd = 8  callref
= 0x80C2
Channel ID i = 0xA9839F
Exclusive, Channel 31
*Oct 11 11:48:24.167: ISDN Se0/2/0:15 Q931: RX - CALL_PROC pd = 8  callref
= 0x80C2
*Oct 11 11:48:31.127: ISDN Se0/2/0:15 Q931: RX - ALERTING pd = 8  callref
= 0x80C2
Progress Ind i = 0x8488 - In-band info or appropriate now available
*Oct 11 11:48:39.975: ISDN Se0/2/0:15 Q931: RX - CONNECT pd = 8  callref =
0x80C2
Connected Number i = '!', 0x83, 'y'
*Oct 11 11:48:39.975: ISDN Se0/2/0:15 Q931: TX - CONNECT_ACK pd = 8
 callref = 0x00C2
*Oct 11 11:49:06.119: ISDN Se0/2/0:15 Q931: TX - DISCONNECT pd = 8
 callref = 0x00C2
Cause i = 0x8090 - Normal call clearing
*Oct 11 11:49:06.423: ISDN Se0/2/0:15 Q931: RX - RELEASE pd = 8  callref =
0x80C2
*Oct 11 11:49:06.423: ISDN Se0/2/0:15 Q931: TX - RELEASE_COMP pd = 8
 callref = 0x00C2

Is there anything I can do from my side to lower this delay?

best regards,

Abebe
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Re: [cisco-voip] ISDN call takes long time before connecting

[Abebe Amare]

I tried putting trailing # but it didn't solve the issue.

Regards

Abebe
On Oct 14, 2014 10:19 PM, Gary Parker g.j.par...@lboro.ac.uk wrote:


 On 14 Oct 2014, at 18:49, Bill Paris bpa...@dncinc.com wrote:

  It sounds like Bell is waiting for more digits. Sending a # after
 sending the number may resolve this issue.

 I was thinking the same thing. That figure of 11 seconds immediately
 jumped out at me as being the default inter-digit timeout on CUCM.

 ---
 /-Gary Parker--f--\
 | Unified Communications Service Manager  |
 n   Loughborough University IT Services   |
 | Tel: +441509635635  Mob: +447989172258  o
 | http://delphium.lboro.ac.uk/pubkey.txt  |
 \r--d-/


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[cisco-voip] Jabber for Windows Contacts

[Abebe Amare]

HI,

I am having my first encounter  with Jabber for windows and it does not
show contacts unless manually added on the client. CUCM is LDAP integrated
with AD for sync and authentication. Is there any config to be done on
CUCM/CUPS side to have all the users in the domain (500) to show up as
contacts?

best regards,

Abebe
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Re: [cisco-voip] Jabber for Windows Contacts

[Abebe Amare]

Hi Mathew

The users can be searched from directory and add manually. The requirement
was for all users to show up as contacts same as they appear in outlook
address book.
Yes I would also agree that having all users as contacts is not productive.

Thanks for your pointers and support.

Best regards

Abebe
On Sep 4, 2014 2:46 PM, Matthew Loraditch 
mloradi...@heliontechnologies.com wrote:

  Are you saying they don’t show up in the search box or they don’t show
 up as actual contacts?

 If the former it appears you have a problem with your directory settings,
 what are you using, UDS or LDAP?

 If the latter, you could bulk import them in IMP but I’m not sure why
 you’d want to do that.

 People generally use contacts as their personal list of folks they want to
 have listed for east of access. 500 contacts would make the contact list
 incredibly busy IMO and unwieldy for finding the people you actually want.





 Matthew G. Loraditch – CCNP-Voice, CCNA-RS, CCDA

 1965 Greenspring Drive
 Timonium, MD 21093

 direct voice. 443.541.1518
 fax.  410.252.9284

 Twitter http://twitter.com/heliontech  |  Facebook
 http://www.facebook.com/#!/pages/Helion/252157915296  | Website
 http://www.heliontechnologies.com/  |  Email Support
 supp...@heliontechnologies.com?subject=Technical%20Support%20Request

 Support Phone. 410.252.8830





 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
 Of *Abebe Amare
 *Sent:* Thursday, September 04, 2014 3:02 AM
 *To:* cisco voip
 *Subject:* [cisco-voip] Jabber for Windows Contacts



 HI,



 I am having my first encounter  with Jabber for windows and it does not
 show contacts unless manually added on the client. CUCM is LDAP integrated
 with AD for sync and authentication. Is there any config to be done on
 CUCM/CUPS side to have all the users in the domain (500) to show up as
 contacts?



 best regards,



 Abebe

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Re: [cisco-voip] Jabber for Windows Contacts

[Abebe Amare]

Hi Philip,

Thanks for the colurful explanation.
On Sep 4, 2014 3:07 PM, Walenta, Philip philip.wale...@polycom.com
wrote:

 It should be noted that this is the way most IMP systems work.  You
 definitely wouldn't want all possible Google contacts showing up in your
 list automatically :)

 Sent from my iPhone

 On Sep 4, 2014, at 6:54 AM, Abebe Amare abu...@gmail.com wrote:

 Hi Mathew

 The users can be searched from directory and add manually. The requirement
 was for all users to show up as contacts same as they appear in outlook
 address book.
 Yes I would also agree that having all users as contacts is not productive.

 Thanks for your pointers and support.

 Best regards

 Abebe
 On Sep 4, 2014 2:46 PM, Matthew Loraditch 
 mloradi...@heliontechnologies.com wrote:

  Are you saying they don’t show up in the search box or they don’t show
 up as actual contacts?

 If the former it appears you have a problem with your directory settings,
 what are you using, UDS or LDAP?

 If the latter, you could bulk import them in IMP but I’m not sure why
 you’d want to do that.

 People generally use contacts as their personal list of folks they want
 to have listed for east of access. 500 contacts would make the contact list
 incredibly busy IMO and unwieldy for finding the people you actually want.





 Matthew G. Loraditch – CCNP-Voice, CCNA-RS, CCDA

 1965 Greenspring Drive
 Timonium, MD 21093

 direct voice. 443.541.1518
 fax.  410.252.9284

 Twitter http://twitter.com/heliontech  |  Facebook
 http://www.facebook.com/#!/pages/Helion/252157915296  | Website
 http://www.heliontechnologies.com/  |  Email Support
 supp...@heliontechnologies.com?subject=Technical%20Support%20Request

 Support Phone. 410.252.8830





 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On
 Behalf Of *Abebe Amare
 *Sent:* Thursday, September 04, 2014 3:02 AM
 *To:* cisco voip
 *Subject:* [cisco-voip] Jabber for Windows Contacts



 HI,



 I am having my first encounter  with Jabber for windows and it does not
 show contacts unless manually added on the client. CUCM is LDAP integrated
 with AD for sync and authentication. Is there any config to be done on
 CUCM/CUPS side to have all the users in the domain (500) to show up as
 contacts?



 best regards,



 Abebe

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[cisco-voip] CUCM LDAP integration

[Abebe Amare]

Hi,

I have a CUCM cluster with locally configured users. This was done because
at the time of install the domain infrastructure (Microsoft AD) was not
ready.
The user id filed in CUCM is the IP Phone DN and all users (including Call
Center agents) are using Extension Mobility. In AD the username is the same
as email ID.
What will be the best way to integrate CUCM with AD for LDAP sync and
authentication?

best regards,

Abebe
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Re: [cisco-voip] CUCM LDAP integration

[Abebe Amare]

Thanks Mathew for the tip.

The main reason behind the push for LDAP integration is to deploy Cisco
Jabber for everyone and I was thinking to use domain ID to login to jabber.
The number of users is not big (~500).
 On Aug 28, 2014 5:40 PM, Matthew Loraditch 
mloradi...@heliontechnologies.com wrote:

  When setting up your integration select the telephone number field to be
 used as User ID.





 Matthew G. Loraditch – CCNP-Voice, CCNA-RS, CCDA

 1965 Greenspring Drive
 Timonium, MD 21093

 direct voice. 443.541.1518
 fax.  410.252.9284

 Twitter http://twitter.com/heliontech  |  Facebook
 http://www.facebook.com/#!/pages/Helion/252157915296  | Website
 http://www.heliontechnologies.com/  |  Email Support
 supp...@heliontechnologies.com?subject=Technical%20Support%20Request

 Support Phone. 410.252.8830





 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
 Of *Abebe Amare
 *Sent:* Thursday, August 28, 2014 10:26 AM
 *To:* cisco voip
 *Subject:* [cisco-voip] CUCM LDAP integration



 Hi,



 I have a CUCM cluster with locally configured users. This was done because
 at the time of install the domain infrastructure (Microsoft AD) was not
 ready.

 The user id filed in CUCM is the IP Phone DN and all users (including Call
 Center agents) are using Extension Mobility. In AD the username is the same
 as email ID.

 What will be the best way to integrate CUCM with AD for LDAP sync and
 authentication?



 best regards,



 Abebe

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Re: [cisco-voip] CUCMBE6K RAM requirement

[Abebe Amare]

I opened the top cover and found that it contains the 4x UC-MR-1X082RY-A,
for some reason it was not shown in the BOQ I was given.

Thanks everyone for the support, greatly appreciated

Abebe




On Wed, May 28, 2014 at 11:41 PM, Ryan Ratliff (rratliff) 
rratl...@cisco.com wrote:

  The server should have come with (4) of the UC-MR-1X082RY-A part for a
 total of 32GB of RAM.

  If it didn't talk to your vendor or Cisco Customer Service so the issue
 can be corrected.

 -Ryan

  On May 28, 2014, at 10:24 AM, Abebe Amare abu...@gmail.com wrote:

  Hi Mathew,

  I will inspect the internal of the server physically to check whether it
 has 8GB or 32 GB.

  Thanks for the support, much appreciated.

  Abebe


 On Wed, May 28, 2014 at 4:56 PM, Matthew Loraditch 
 mloradi...@heliontechnologies.com wrote:

  If that is the part that is ordered all that is, is the version w/o
 strong encryption for certain countries. You should have 32GBs of RAM. If
 you don’t you have a problem and should be able to correct with your Disti
 or TAC as you got a bad product.

 Have you spun up the server and it actually only shows 8GBs?



 image001.jpg

 Matthew G. Loraditch – CCNP-Voice, CCNA-RS, CCDA

 1965 Greenspring Drive
 Timonium, MD 21093

 direct voice. 443.541.1518
 fax.  410.252.9284

 Twitter http://twitter.com/heliontech  |  Facebook
 http://www.facebook.com/#!/pages/Helion/252157915296  | Website
 http://www.heliontechnologies.com/  |  Email Support
 supp...@heliontechnologies.com?subject=Technical%20Support%20Request

 Support Phone. 410.252.8830

 image002.png



 *From:* Abebe Amare [mailto:abu...@gmail.com]
 *Sent:* Wednesday, May 28, 2014 9:48 AM
 *To:* Matthew Loraditch
 *Cc:* cisco voip
 *Subject:* Re: [cisco-voip] CUCMBE6K RAM requirement



 Hi Mathew,



 The full order is spec is as below:

 *Part Number*

 *Description*

 BE6K-ST-BDL-XU=

 Cisco BE6000 UCS C220M3 MD Srv,UNRST 9.x SW,Hyp,UPM,VCS

 LIC-VCS-10+

 Video Comm Server 10 Add Non-traversal Network Calls

 CAB-9K10A-EU

 Power Cord, 250VAC 10A CEE 7/7 Plug, EU

 BE6K-SW-APP-9.X-XU

 Cisco Business Edition 6000 SW App Version 9.X Unrestricted

 CIT-PSU-BLKP

 Power Supply Blanking Panel/Filler

 CIT-SD-16G-C220

 16GB SD Card Module for C220 servers

 CTI-VCSC-BE6K-PAK

 Config Only E-Delivery VCS Control PAK PID

 VMW-VS5-HYP-K9

 Cisco UC Virt. Hypervisor 5.x (2-socket)

 VMW-VS5-SNS

 Cisco UC Virt. Hypervisor 5.x - SnS

 LIC-VCS-BASE

 License Key - VCS Non-encrypted Software Image

 LIC-VCS-GW

 Enable GW Feature (H323-SIP)

 LIC-VCSE-5+

 Video Communication Server - 5 Traversal Calls

 R2XX-RAID10

 Enable RAID 10 Setting

 UC-A03-D500GC3

 500GB  6Gb SATA  7.2K RPM SFF Hot Plug/Drive Sled Mounted

 UC-CPU-E5-2609

 2.4 GHz E5-2609/80W 4C/10MB Cache/DDR3 1066MHz

 UC-MR-1X082RY-A

 8GB DDR3-1600-MHz RDIMM/PC3-12800/Dual Rank/1.35v

 UC-PSU-650W

 650W Power Supply Unit For UCSC C220 Rack Server

 UC-RAID-9271

 MegaRAID 9271-8i + Battery Backup for C240 and C220



 It looks like the part number they ordered is different to your
 recommendation of BE6K-ST-BDL-k9=







 On Wed, May 28, 2014 at 4:08 PM, Matthew Loraditch 
 mloradi...@heliontechnologies.com wrote:

  If they bought the correct BE6K server bundle it would have come with
 the right RAM. It sounds like they got the wrong thing. I’d advise them to
 RMA and order the BE6K-ST-BDL-K9= part as I’m guessing they purchased a
 build to order C220 M3 and it’s not going to have come with the right
 entitlements or base software and is not discounted the same way as the
 bundle part.





 Matthew G. Loraditch – CCNP-Voice, CCNA-RS, CCDA

 1965 Greenspring Drive
 Timonium, MD 21093

 direct voice. 443.541.1518
 fax.  410.252.9284

 Twitter http://twitter.com/heliontech  |  Facebook
 http://www.facebook.com/#!/pages/Helion/252157915296  | Website
 http://www.heliontechnologies.com/  |  Email Support
 supp...@heliontechnologies.com?subject=Technical%20Support%20Request

 Support Phone. 410.252.8830





 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On
 Behalf Of *Abebe Amare
 *Sent:* Wednesday, May 28, 2014 8:53 AM
 *To:* cisco voip
 *Subject:* [cisco-voip] CUCMBE6K RAM requirement



 Hi All,



 Customer has a single CUCMBE6K with 8GB of RAM (UCS C220M3 medisum
 density Server).

 . They want to install on it CUCM, UC, Presence and Informacast.
 According to the docwiki for BE6k implementation, for Medium Density
 (MD) server, co-residency policy supports for any 4 UC applications and 1
 Provisioning application as long as they fit in 8vCPU and 32GvRAM. Does
 this mean they need to upgrade the RAM to 32 GB to accommodate all the
 servers?



 Thanks in advance,



 Abebe




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Re: [cisco-voip] CUCMBE6K RAM requirement

[Abebe Amare]

Hi Mathew,

The full order is spec is as below:

*Part Number*

*Description*

BE6K-ST-BDL-XU=

Cisco BE6000 UCS C220M3 MD Srv,UNRST 9.x SW,Hyp,UPM,VCS

LIC-VCS-10+

Video Comm Server 10 Add Non-traversal Network Calls

CAB-9K10A-EU

Power Cord, 250VAC 10A CEE 7/7 Plug, EU

BE6K-SW-APP-9.X-XU

Cisco Business Edition 6000 SW App Version 9.X Unrestricted

CIT-PSU-BLKP

Power Supply Blanking Panel/Filler

CIT-SD-16G-C220

16GB SD Card Module for C220 servers

CTI-VCSC-BE6K-PAK

Config Only E-Delivery VCS Control PAK PID

VMW-VS5-HYP-K9

Cisco UC Virt. Hypervisor 5.x (2-socket)

VMW-VS5-SNS

Cisco UC Virt. Hypervisor 5.x - SnS

LIC-VCS-BASE

License Key - VCS Non-encrypted Software Image

LIC-VCS-GW

Enable GW Feature (H323-SIP)

LIC-VCSE-5+

Video Communication Server - 5 Traversal Calls

R2XX-RAID10

Enable RAID 10 Setting

UC-A03-D500GC3

500GB  6Gb SATA  7.2K RPM SFF Hot Plug/Drive Sled Mounted

UC-CPU-E5-2609

2.4 GHz E5-2609/80W 4C/10MB Cache/DDR3 1066MHz

UC-MR-1X082RY-A

8GB DDR3-1600-MHz RDIMM/PC3-12800/Dual Rank/1.35v

UC-PSU-650W

650W Power Supply Unit For UCSC C220 Rack Server

UC-RAID-9271

MegaRAID 9271-8i + Battery Backup for C240 and C220

It looks like the part number they ordered is different to your
recommendation of BE6K-ST-BDL-k9=




On Wed, May 28, 2014 at 4:08 PM, Matthew Loraditch 
mloradi...@heliontechnologies.com wrote:

  If they bought the correct BE6K server bundle it would have come with
 the right RAM. It sounds like they got the wrong thing. I’d advise them to
 RMA and order the BE6K-ST-BDL-K9= part as I’m guessing they purchased a
 build to order C220 M3 and it’s not going to have come with the right
 entitlements or base software and is not discounted the same way as the
 bundle part.





 Matthew G. Loraditch – CCNP-Voice, CCNA-RS, CCDA

 1965 Greenspring Drive
 Timonium, MD 21093

 direct voice. 443.541.1518
 fax.  410.252.9284

 Twitter http://twitter.com/heliontech  |  
 Facebookhttp://www.facebook.com/#!/pages/Helion/252157915296
 | Website http://www.heliontechnologies.com/  |  Email 
 Supportsupp...@heliontechnologies.com?subject=Technical%20Support%20Request

 Support Phone. 410.252.8830





 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
 Of *Abebe Amare
 *Sent:* Wednesday, May 28, 2014 8:53 AM
 *To:* cisco voip
 *Subject:* [cisco-voip] CUCMBE6K RAM requirement



 Hi All,



 Customer has a single CUCMBE6K with 8GB of RAM (UCS C220M3 medisum
 density Server).

 . They want to install on it CUCM, UC, Presence and Informacast. According
 to the docwiki for BE6k implementation, for Medium Density (MD) server,
 co-residency policy supports for any 4 UC applications and 1 Provisioning
 application as long as they fit in 8vCPU and 32GvRAM. Does this mean they
 need to upgrade the RAM to 32 GB to accommodate all the servers?



 Thanks in advance,



 Abebe

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[cisco-voip] PVDM

[Abebe Amare]

Hi everyone,

The following is a snippet from the detail order for Cisco 2901 voice
router:

1. C2901-CME-SRST/K9 - 2901 Voice Bundle w/ PVDM3-16,FL-CME-SRST-25, UC
License PAK
2. PVDM3-16U32 - PVDM3 16-channel to 32-channel factory upgrade

What does the PVDM3-16U32 mean? is it a separate PVDM or license? also if I
want to purchase a VWIC3-2MFT-T1/E1, what is the PVDM3 type I have to get?
(I dont have the right login credentials to use DSP calculator)

best regards,

Abebe
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Re: [cisco-voip] PVDM

[Abebe Amare]

Thanks James for the explanation.
From your reply I understand that I have a 32 channel DSP on the
PVDM3-16U32. If I purchase a PVDM3-64, on top of the existing one, I will
have 96 channels. Is that correct?

best regards,

Abebe

On Wed, May 21, 2014 at 10:22 AM, James Buchanan
james.buchan...@gmail.comwrote:

   Hello,

 They are taking the 16-channel DSP and upgrading it to 32-channels.

 The number of DSP channels you need depending on how many channels of the
 PRI you intend to use, plus conferencing and transcoding. This particular
 VWIC card can do 60 channels as an E1 and 48 channels as a T1. Do you
 intend to configure two PRIs? If so, you need one DSP channel per T1/E1
 channel as a minimum. I would recommend upgrading the 32 channels to a
 minimum of 96 channels so that you have enough resources for both T1/E1
 ports plus transcoding and conferencing.

 Now, the number of DSP channels also depends on what codec you intend to
 use. For example G.729 is a high-density codec, meaning that more DSP
 channels might be required for transcoding. G.729b does not use as many DSP
 channels.

 I hope this helps!

 James


  On Wed, May 21, 2014 at 9:14 AM, Abebe Amare abu...@gmail.com wrote:

  Hi everyone,

 The following is a snippet from the detail order for Cisco 2901 voice
 router:

 1. C2901-CME-SRST/K9 - 2901 Voice Bundle w/ PVDM3-16,FL-CME-SRST-25, UC
 License PAK
 2. PVDM3-16U32 - PVDM3 16-channel to 32-channel factory upgrade

 What does the PVDM3-16U32 mean? is it a separate PVDM or license? also if
 I want to purchase a VWIC3-2MFT-T1/E1, what is the PVDM3 type I have to
 get? (I dont have the right login credentials to use DSP calculator)

 best regards,

 Abebe



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Re: [cisco-voip] PVDM

[Abebe Amare]

Thanks James for your support.

best regards,

Abebe

On Wed, May 21, 2014 at 10:57 AM, James Buchanan
james.buchan...@gmail.comwrote:

 Exactly. However, you may want to price adding the 64-channel versus
 upgrading the 16 to a 64 and adding a 32. It'll probably be close in price,
 but just compare. An alternative would be to just upgrading the 16 to a
 128, and that would leave you a free slot for more DSPs in the future.


 On Wed, May 21, 2014 at 9:53 AM, Abebe Amare abu...@gmail.com wrote:

 Thanks James for the explanation.
 From your reply I understand that I have a 32 channel DSP on the
 PVDM3-16U32. If I purchase a PVDM3-64, on top of the existing one, I will
 have 96 channels. Is that correct?

 best regards,

 Abebe

 On Wed, May 21, 2014 at 10:22 AM, James Buchanan 
 james.buchan...@gmail.com wrote:

   Hello,

 They are taking the 16-channel DSP and upgrading it to 32-channels.

 The number of DSP channels you need depending on how many channels of
 the PRI you intend to use, plus conferencing and transcoding. This
 particular VWIC card can do 60 channels as an E1 and 48 channels as a T1.
 Do you intend to configure two PRIs? If so, you need one DSP channel per
 T1/E1 channel as a minimum. I would recommend upgrading the 32 channels to
 a minimum of 96 channels so that you have enough resources for both T1/E1
 ports plus transcoding and conferencing.

 Now, the number of DSP channels also depends on what codec you intend to
 use. For example G.729 is a high-density codec, meaning that more DSP
 channels might be required for transcoding. G.729b does not use as many DSP
 channels.

 I hope this helps!

 James


  On Wed, May 21, 2014 at 9:14 AM, Abebe Amare abu...@gmail.com wrote:

  Hi everyone,

 The following is a snippet from the detail order for Cisco 2901 voice
 router:

 1. C2901-CME-SRST/K9 - 2901 Voice Bundle w/ PVDM3-16,FL-CME-SRST-25,
 UC License PAK
 2. PVDM3-16U32 - PVDM3 16-channel to 32-channel factory upgrade

 What does the PVDM3-16U32 mean? is it a separate PVDM or license? also
 if I want to purchase a VWIC3-2MFT-T1/E1, what is the PVDM3 type I have to
 get? (I dont have the right login credentials to use DSP calculator)

 best regards,

 Abebe



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Re: [cisco-voip] NTP Issue

[Abebe Amare]

I have encountered such issue on 7941 phones mostly but only a few of them
behave like this. Factory reset solves the problem for me.

Regards,

Abebe
On Mar 24, 2014 3:33 PM, Jason Aarons (AM) jason.aar...@dimensiondata.com
wrote:

 You can http to the phone ip address and check it's time/timezone, etc.



 *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf
 Of *Matthew Collins
 *Sent:* Monday, March 24, 2014 7:53 AM
 *To:* costas georgiou; cisco-voip@puck.nether.net
 *Subject:* Re: [cisco-voip] NTP Issue





 Hi Costas,



 I take it you are using 89XX phones?



 CSCun11142 is a bug, affecting 89XX phones daylight settings.



 Usually daylight saving is 4th Sunday of march, But this year the last
 Sunday is the 5th Sunday.



 Regards



 Matthew





 *From:* cisco-voip 
 [mailto:cisco-voip-boun...@puck.nether.netcisco-voip-boun...@puck.nether.net]
 *On Behalf Of *costas georgiou
 *Sent:* 24 March 2014 11:25
 *To:* cisco-voip@puck.nether.net
 *Subject:* [cisco-voip] NTP Issue



 Hi All,

 I have a CUCM running 8.6.2, I have been informed this morning that some
 IP phones are 1 hr out of sync. This is happening to a few remote sites.
 The NTP server is seperate and is looked after by the customer.

 I have check the CUCM server and get the follwing results which look ok:


 admin:utils ntp status

 ntpd (pid 26786) is running...



  remote   refid  st t when poll reach   delay   offset
  jitter


 ==

 +10.160.x.x.10.107.195.1 4 u  572 1024  377   16.7430.105
 0.015

 *165.72.x.x.GPS.1 u  659 1024  377   35.434   -0.071
 7.146





 synchronised to NTP server (165.72.x.x) at stratum 2

time correct to within 32 ms

polling server every 1024 s



 Current time in UTC is : Mon Mar 24 09:36:02 UTC 2014

 Current time in Europe/London is : Mon Mar 24 09:36:02 GMT 2014

 Admin:



 The clock on the routers sjow the same time.  Any ideas what could be the
 problem?



 Thanks in advance for your help.



 Regards



 Costas









 itevomcid

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