On Thu, Jan 30, 2014 at 8:48 AM, Vignesh Sethuraman
sethuvign...@gmail.comwrote:
Media Termination Point Required (Checked)
MTP Preferred Originating CodecRequired Field: g711ulaw
Hi Vignesh,
I think if you can set these two to default settings which is MTP Required
[uncheck], and MTP
Hi Vignesh,
Would it be possible to make a test call from PSTN phone too?Is the
result different than call made from SiteB PH2/PH3?
Also it might be worth checking the dtmf-relay settings on relevant VoIP
dial-peer(s) on SiteA GW too.
Regards,
--Somphol.
On Wed, Jan 22, 2014 at 9:49 PM,
On Fri, Nov 15, 2013 at 9:14 AM, Olusegun Oguntuga
segunogunt...@gmail.comwrote:
Can anyone please explain what exactly needs to be done to get calling
name displayed on an enterprise phone when a call is a received via mobile
voice access with inbound fast start enabled.
Hi Olusegun,
I
Hi StefanoS,
The typical CUCM SDI's DETAIL trace with default ticks should be enough.
I think it is likely that the CSS applied to RDP doesn't have access to
your Mobile Voice Access Directory Number' media resource. (The one you
define in Media Resources - Mobile Voice Access),I found that
...@hotmail.com
CC: ccie_voice@onlinestudylist.com
It is also very good to see lab slots available on Saturday and Sunday at
both RTP San Jose. Weekend lab seem to be for those two locations only
at the moment.
--Somphol.
On Thu, Oct 31, 2013 at 8:43 AM, Somphol Boonjing somp...@gmail.com
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
On Sun, Nov 3, 2013 at 2:02 AM, Pavan K pav.c...@gmail.com wrote:
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
On Thu, Oct 31, 2013 at 8:21 AM, Bill Tolentino btolent...@hotmail.comwrote:
Then at approximately 7:10am they were all gone. I have been re-checking
all day and still no dates for any sites.?
Hi Bill,
I have just checked.There are still available slots in Bangalore. 17
in Nov/ 6 in
It is also very good to see lab slots available on Saturday and Sunday at
both RTP San Jose. Weekend lab seem to be for those two locations only
at the moment.
--Somphol.
On Thu, Oct 31, 2013 at 8:43 AM, Somphol Boonjing somp...@gmail.com wrote:
On Thu, Oct 31, 2013 at 8:21 AM, Bill
Just to share some findings on BACD experiment.
My BACD is both for the embedded BACD and the external TCL-based BACD
(2.1.x.x) running on IOS 12.4(15)T.
I always think BACD is fairly straightforward and well-document.And, I
have never come close to question the validity of Cisco's own BACD
Hi StefanoS,
Just a tiny addition on the good collection of the link you've gathered,
with the long navigation to Cisco IP Phone customization, you may want to
get the the template from
The url directories under enterprise parameters
There you will see a URL that you can use to retrieve a
The one thing I'm really struggling with is mapping out my dial-plan
during my read through of the lab. I would love to hear what others are
doing.
In my previous attempts, I find it very hard too, because the questions are
verbose and I could either spend too much time reading OR not able to
better than on paper. I make
to many mistakes on paper and can hardly read what I wrote. Thanks!!
On Fri, Oct 18, 2013 at 2:53 AM, Somphol Boonjing somp...@gmail.comwrote:
The one thing I'm really struggling with is mapping out my dial-plan
during my read through of the lab. I would love
+1 for that. Awesome information. Imagine this come out in one of the
next revision of the exam on backup Route List and on purpose remove the
ability to change this parameter on Call Manager. (4 points)!!!
Thank you very much for sharing.
Regards,
--Somphol.
On Sat, Oct 19, 2013 at 3:10
Hi,
On service parameters, you may also want to check Vik's article
http://blog.ipexpert.com/2010/10/13/common-ucm-service-parameters-to-change/
.
On the comment section, Trinifox also mentioned Please add: Intraregion
Audio Codec Default to G729 to avoid CSCsl74701 Bug.
In my checklist, I also
Sorry for revisiting this old thread. The Calling Party Transformation at
the Device Pool level would come in handy for this particular need.
In the document starting 7.1.2, this is stated explicitly,
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_1_2/ccmfeat/fscallpn.html#wp1325305
Hi All,
Am I understand correctly that unlike voice translation profile on IOS
gateway, the calling party translation pattern, that is applied to gateway
level for outgoing call, can't be tailored based on destination route
pattern?
For example, assuming both Site B and Site C are in SRST mode,
Products - Voice and Unified Communications - Unified Communications
Applications - Unified Communications Clients - Cisco Unified Personal
Communicator
Look at Release Notes
http://www.cisco.com/en/US/docs/voice_ip_comm/cupc/7_1/english/release/notes/cupc71.html
Regards,
--Somphol.
On Tue,
Hi Hesham,
On Wed, Sep 4, 2013 at 4:23 AM, Hesham Abdelkereem heshamcentr...@gmail.com
wrote:
the number of concurrent calls on each PRI for example today from 8am to
5PM.
I played around with SNMP to collect that values for a while. I remember
that there is no MIBS OID for concurrent
Hi Sam,
I think I came across similar error message. You may want to adjust your
Voice-Sig class-map as following:
!
class-map match-any Voice-Sig
match ip dscp cs3 af31 === list them in the same line
class-map match-any Voice-RTP
match ip dscp ef
!
On Thu, Aug 29, 2013 at 4:47 PM, Sam
You need to put in extra layer of policy-map for this.
class-map match-any RTP
match protocol rtp
class-map match-any SIG
match protocol skinny
match protocol sip
match protocol h323
match protocol custom-01
!
policy-map LLQ
class SIG
bandwidth
PM, Somphol Boonjing somp...@gmail.comwrote:
This might be worth revisiting.Forgive me if this is not entirely a
new insight.
In short, be aware that as soon as the command service-policy input XXX
in entered into the configuration, the mls qos trust cos/dscp will be
removed. Likewise
-SC
Regards,
--Somphol.
On Tue, Aug 6, 2013 at 6:16 PM, Somphol Boonjing somp...@gmail.com wrote:
Hi,
Can anyone help confirm my understanding on this topic?
My observation is that Per VC fragmentation, while it can be configured as
when in the example below, is not very useful
This might be worth revisiting.Forgive me if this is not entirely a new
insight.
In short, be aware that as soon as the command service-policy input XXX
in entered into the configuration, the mls qos trust cos/dscp will be
removed. Likewise, if the command mls qos trust cos/dscp is
This question is definitely one of those, one that seems very simple either
to confirm or deny. Either Yes it is 100% support or No it is 100%
not. But googling around, and it is very confusing. Part of it may be
because this problem only happens to in-call via Mobile Voice Access IVR.
(The
On Sun, Aug 11, 2013 at 11:17 AM, Somphol Boonjing somp...@gmail.comwrote:
Reading through two thread originated by a candidate by the name
datucha/datoc, read through it both discussion threads, you will see
how confusing this can be.
http://ieoc.com/forums/p/18782/162049.aspx
http
On Fri, Aug 9, 2013 at 1:15 PM, Alex Mendoza aa.mend...@icloud.com wrote:
Calling from my SNR to MVA is working, MVA asks for my pin number, then
press 1, after that I dialed internal 4 digit extension but this internal
phone only shows the caller number and not the caller name.
I think is
Hi,
Can anyone help confirm my understanding on this topic?
My observation is that Per VC fragmentation, while it can be configured as
when in the example below, is not very useful if not configured for all of
the existing PVC that shared the same physical interface, isn't it?
With the example
Hi,
Forgive me to chime in and most likely this won't answer the original
question outright.
Hesham's response is fairly inspiring in a way. It inspires me think a
lot harder about this area.
- Take another closer look at what is available, it is intriguing to see
Layer 3 information such as
Hi Hesham,
My first guess would be an extra DTMF somehow. You application is running
correctly, but I seems to think that it has received an unintentional
digit.
I would try to isolate the problem first by make a POTS dial-peer and test
call from PSTN.If the problem still persists, you may
dial-peer voice 4001 pots
service app-b-acd-aa
incoming called-number *4008*
!
I mean a test call from PSTN to x*4008*.
--Somphol.
On Tue, Jul 2, 2013 at 10:15 PM, Somphol Boonjing somp...@gmail.com wrote:
Hi Hesham / Khaled,
Fully agreed with that Kaled on that typo.
Just a few more
Hi Aman,
In case this help, the topic seems to be discussed before.
E.g. in the following thread.
- http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg06632.html
- http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg31952.html
Regards,
--Somphol.
On Fri, Jun 28, 2013 at
Hi Aman,
But voice packets font get fragmented when you use frf.12. So why would
you use 8 and not 4?
Just to try to dig up some relevant information.
*[1] On whether voice packet get fragmented.*
I agreed fully with you. If configured correctly, the voice packet should
not be fragemented.
Frame Relay Header than
4 bytes.
Regards,
--Somphol.
On Fri, Jun 28, 2013 at 6:39 PM, Somphol Boonjing somp...@gmail.com wrote:
Hi Aman,
But voice packets font get fragmented when you use frf.12. So why
would you use 8 and not 4?
Just to try to dig up some relevant information.
*[1
This may have nothing to do with the implementation checklist, but may be
useful as part of a verification steps. Assuming SiteA SiteC are
configured with a typical scenario with G711 inter-region and G729
intra-region, and RSVP is required.
1. Incoming or outgoing PSTN call to or from say SC
- XCODER: XCODER + RSVP
(may i know what is purpose)
- if they ask codec G711, we should see 80k insh rsvp reservation ?
tks
K
*From:* Somphol Boonjing somp...@gmail.com
*To:* Karen Johnson karen.johnson...@yahoo.ca
*Cc:* ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Hi Hesham,
knowing that Gatekeeper is working with SiteB under normal operation but
doesn't work with CFUR
Could you please clarify the problem you are facing? What do you mean
when you say the gatekeeper is not working with CFUR?
Any Ideas,
I think we will need to simplify the scenario to
to be registered to CUCM rather than
CALL MANAGER FALLBACK
the call go through via Gatekeeper
Many Thanks,
Hesham
On 22 June 2013 23:26, Somphol Boonjing somp...@gmail.com wrote:
Hi Hesham,
knowing that Gatekeeper is working with SiteB under normal operation
but doesn't work with CFUR
voice-mail 5003
!
dial-peer voice 222 voip
service app-b-acd-aa
destination-pattern 8005550123
session target ipv4:192.168.1.1
incoming called-number 8005550123
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
On Sat, Jun 22, 2013 at 2:25 AM, Somphol Boonjing somp
help.
On the H323 CUCM Trunk, RTMT Real Time monitoring with Detailed Debug
turn on would help you see whether the H323 Trunk has the right CSS to
reach x3001.
Hope this gives you some idea to work on this case.
Regards,
--Somphol.
On Sun, Jun 23, 2013 at 5:27 PM, Somphol Boonjing somp
that will show me the gatekeeper
call flow.
I have been a long time never worked with that.
Thanks for your ideas,
I will keep you and the forum posted if I got any updates,
Thanks,
Hesham
On 23 June 2013 01:40, Somphol Boonjing somp...@gmail.com wrote:
Hi Hesham,
I have a few ideas. I
all just the CUCM Trunk and has both 2XXX and 3XXX
I think that could make it work
Thank you very much for ur great input
I will test it and let u know
Thank you very much for ur great efforts.
On Jun 23, 2013, at 3:30 AM, Somphol Boonjing somp...@gmail.com wrote:
Hi Hesham
:43 PM, Somphol Boonjing somp...@gmail.com wrote:
Hi Hesham,
Essentially, the gw-priority is to advise the gatekeeper to choose SBGW
over CUCMTRUNK. The higher the number, the higher the priority. Without
this it will distribute the call to 3XXX to both CUCMTRUNK and SBGW in a
round robin
Scripts: Example
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html
On 22/06/2013 4:18 PM, CISCO CCIE VOICE ccievoic...@gmail.com wrote:
application
no service app-b-acd
no service app-b-acd-aa
On Sat, Jun 22, 2013 at 9:06 AM, Somphol Boonjing somp
:
*CUCM service Parameter : Allow Peer to Preserve H.323 Calls*
*
*
Keep it coming guys and gals
Regards,
Ovidiu
On Fri, Jun 21, 2013 at 11:55 AM, Somphol Boonjing somp...@gmail.comwrote:
To add to your list that is already good,
- date/time format
- timezone
- system message
A few points that I think worth double check:
[1]
Assuming this is the configuration of the DHCP on CUCM, I think the primary
router doesn't require subnet mask to be specified
DHCP Server : 10.10.210.10
subnet IPV4 address: 10.10.200.0
primary start addr: 10.10.200.120
primary end addr :
-b-acd-aa
destination-pattern 8005550123
session target ipv4:192.168.1.1
incoming called-number 8005550123
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
On Sat, Jun 22, 2013 at 2:25 AM, Somphol Boonjing somp...@gmail.comwrote:
That one is the embedded one so you actually
Somphol, I will start debugging at the router level.
The SVI was initially not there, I was trying few different things to
get this to work added it.
On Sat, Jun 22, 2013 at 5:36 AM, Somphol Boonjing somp...@gmail.comwrote:
A few points that I think worth double check:
[1]
Assuming
To add to your list that is already good,
- date/time format
- timezone
- system message
- number of max calls busy call triggers
- call pickup behavior if applicable (directed vs no directed call pickup)
- call-transfer pattern
- call-forward pattern
- number of channels for ephone (dual,
://www.cisco.com/en/US/docs/voice_ip_comm/cucm/port/8_5_1/portlist851.html
On Fri, Jun 7, 2013 at 6:50 PM, Somphol Boonjing somp...@gmail.com
wrote:
I love the detail step that Bill outlined above.+1 for that.
The other day I remember seeing someone mentioned something about
Hi,
I think the easiest way is to check UCCX Service Parameters under System
menu.
--Somphol.
--Somphol
On Sun, Jun 16, 2013 at 7:36 PM, CISCO CCIE VOICE ccievoic...@gmail.comwrote:
Hi,
How can i verify that UCCX is using G729 codec native
Thanks
Just my personal experience when I failed my first lab attempt. CUE is at
the forefront of the time waster. Simply because it requires constant
reboot if mistake is made and every reboots take a long time too.
10 ) CUE integration and trafer setup takes 20 mins
My brain seems to go numb at
Thanks, Bill for great information.
BTW, if I could ask about your thought on the VNC-only workstation. I
don't really understand the logic behind making and RDP available for
Windows-based UCCX server, and only supply the VNC session for another
utility host. (I was in a rush to such an
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