An easier way to test for THEO and such is to just shut down the voice-port
(not the controller or serial).
Quick and easy and perhaps not as easy to overlook when troubleshooting.
I've left my null routes in a couple of times without realizing it.
From: ccie_voice-boun...@onlinestudylist.com
[m
I've found the QoS questions are very specific to test a certain area of
knowledge. They are not looking for what we would consider a "best practice"
system wide. I think we could skip setting the DSCP values in CUCM.
If you think the question calls for it you can have your class-map match both
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,
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 als
I agree to setting the service parameters to default first. I was planning
on doing that myself. As to changing the DSCP values, it all depends on
what they ask for in the QoS section of the test is all.
On Thu, Oct 17, 2013 at 2:24 PM, probert...@gmail.com
wrote:
> Hi,
>
> I think my strategy
Hi,
I think my strategy will be to set all Service Parameters to default before
making changes. This way I can avoid and undesirable presets.
Let me know your thoughts on this.
Also why are you setting DSCP for Phone Configuration and DSCP for Cisco
CallManager to Device Interface to AF31? Defaul
My test is just a couple of weeks away, and I've been reading different
blogs on how to maximize your time. 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.
I have also been building base router co
Thank you everyone for your help on this. I was bald already, but if I had
hair I would have been pulling it out on this one.
On Thu, Oct 17, 2013 at 11:23 AM, Ramcharan Arya
wrote:
> This is one of the best blog written by Vik on SRST it covers all details.
>
>
>
> http://blog.ipexpert.com/201
HiFor telephony-service I used:srst mode auto-provision allsrst dn line-mode dualAnd for call-manager-fallback I used:max-ephone 4max-dn 8 dual-lineAm I wrong using these configurations?I avoid octo-lines for the bug mentioned.best regards!AlexOn Oct 17, 2013, at 10:18 AM, Ramcharan Arya wrote:Hi
If you are talking about testing redundancy I'll do the same thing on the
gateway I want to simulate as being down. For example when doing TEHO where
if the remote gateway is down we want to fail to the local gateway, I'll go
to the remote gateway and put in the static routes.
On Thu, Oct 17, 201
This is one of the best blog written by Vik on SRST it covers all details.
http://blog.ipexpert.com/2012/03/12/high-availability-series-3-unified-cme-for-srst-gotchas/
Thanks,
Ramcharan Arya CCIE # 28926 (Voice/Routing & Switching)
On Thu, Oct 17, 2013 at 11:05 AM, Martin Sloan wrote:
> Bill
Bill,
Here's a great reference for CME MWI:
http://ciscovoiceguru.com/518/cue-mwi-notification-methods/
I've used this a lot through my studies.
@Ramcharan - good call. I think the symptom with that bug is that the
phones will register but display no DN's and if you issue 'show ephone reg'
it
Ok the re-boot worked and now I can register my phones to CME SRST on a WAN
failure, and the CUE MWI works!!
I did have to go in and edit the call-foward parameters on the ephone-dn
with voicemail, and add my cor-list info to all the lines.
On Thu, Oct 17, 2013 at 10:53 AM, Bill Hatcher wrote:
Ok, I figured out my issue with the MWI not coming on in SRST more. Need
to ass the key word unsolicited to the mwi-server command.
Now to get the CME-SRST working.
On Thu, Oct 17, 2013 at 10:36 AM, Bill Hatcher wrote:
> Seifeddine,
>
> I've run that debug, but there is absolutly no output w
Seifeddine,
I've run that debug, but there is absolutly no output when I'm using
CME-SRST.
Bill
On Thu, Oct 17, 2013 at 10:30 AM, Seifeddine Tlili <
seifeddine.tl...@lvs1.com> wrote:
> Can you send the output of debug ephone register?
>
> ** **
>
> Thx
>
> ** **
>
> *Kindly***
>
> * *
The reason I was even trying to register my phones in CME SRST is begause I
cann't seem to get the CUE MWI to work in SRST and I wanted to see if I was
missing something. I get a SIP 481 Call Leg/Transaction Does Not Exist
when my CME sends mwi. In researching this issue I found that if the voice
Marty,
The weird thing is they work when I use call-manager-fallback. Looking at
the cnf file, all seems well.
On Thu, Oct 17, 2013 at 10:05 AM, Martin Sloan wrote:
> I had a similar issue recently which ended up being a DB replication
> problem. You could check the phones config file:
>
> 10
Ramcharan,
I'll give that a try.
On Thu, Oct 17, 2013 at 10:18 AM, Ramcharan Arya
wrote:
> Hi Bill,
>
> I believe this might be related to a bug with using octo lines in CME SRST.
>
> Come out of SRST and reload the router this might resolve the issue.
>
> Regards,
> Ramcharan Arya CCIE # 28926
Hi Bill,
I believe this might be related to a bug with using octo lines in CME SRST.
Come out of SRST and reload the router this might resolve the issue.
Regards,
Ramcharan Arya CCIE # 28926 ( Voice/Routing & Switching)
On Thu, Oct 17, 2013 at 9:34 AM, Bill Hatcher wrote:
> It’s not workin
I had a similar issue recently which ended up being a DB replication
problem. You could check the phones config file:
10.10.210.11:6970/SEP123456789123.cnf.xml> Check subscribers
copy of the file
Right-click, select view source and search for 'srst' and see what it has
there. I coul
It’s not working!! Can anyone see something I may be doing wrong? My PRI
and CUE register, I can even see SIP MWI being sent, but my phones will not
register. They worked when I was using call-manager-fallback though so I
know my SRST configuration is correct on the CallManager.
telephony-serv
I have been looking for quick and easy ways to test SRST, and I've found
many different waqys of doing this. With the exception of pulling the WAN
interface, they all seem to take a lot of time and effort to accomplish.
Anything from creating access-lists to block the traffic to creating new
call
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