There is no drawback in first option but I would like to be as specific as
possible.
The 2nd rule should be like this:
voice translation-rule 1
rule 1 /^3214\(3...\)$/ /\1/
This way only calls for your DDI range will be entertained by the gateway
and rest will be rejected.
At
By more flexible, you mean, you have less control over it?
I agree with Ashar, more specific is always better.
Use ^ and $ when possible and include the digits you know are not going to
change.
On Fri, Jul 2, 2010 at 1:35 AM, Ashar Siddiqui wrote:
> There is no drawback in first option but I w
Hi All,
Why would my BR1-RTR being registering with UCM on the serial sub interface
10.10.111.2
When i have configured it to use the Loopback0 for source packets..
10.10.110.2
Plus i think is not helping with making any call out of the gateway !!
Thanks
Duncan
_
Try a no mgcp / mgcp on the gateway. When an MGCP router registers with
CUCM for the first time, it will use the serial interface. If it
doesn't work, please send your config.
*Matthew Berry*
*_Vitals:_*
*GVoice: *+1.612.424.5044
*Gmail*: ciscovoiceg...@gmail.com
*Skype*: ciscovoiceguru
isdn switch-type primary-ni
!
ccm-manager switchback immediate
ccm-manager redundant-host 10.10.210.10
ccm-manager mgcp
!
!
controller T1 2/0
framing esf
linecode b8zs
pri-group timeslots 1-3,24 service mgcp
!
!
!
interface Loopback0
ip address 10.10.110.2 255.255.255.25
I have an SRST requirement which states the following:
A. when in SRST phone should should be able to utilize cbarge and Park and
live-record
B. Your are not allowed to allowed to have information for learned phones in
running configuration (no manual ephone configuration)
C. Phones must support
Ok..
Shutdown mgcp
Took out the bind commands ...
Put them back in
No shut on Mgcp
Reloaded
Fixed, now registered with the loopback...calls are flowing !!
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Duncan
Hamilton-Walk
Mgcp / no mgcp is a very important command when dealing with MGCP gateways!
If not using trombone then always do no mgcp/mgcp for any changes you make
at gateway or call manager to come into effect.
Ash>
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylis
I noticed simply turning off mgcp and turning it back on doesnt work for
me.. I usually remove the bind statement from the serial interface.. bounce
it and make sure it comes up then reapply the bind and then turn on mgcp.
___
For more information regardi
Thanks,
But i believe the problem was the fact that the gateway was registering on
the wrong address.. so by removing the bind commands and put they back in..
forced the gateway to register on the correct address.
Thanks
Duncan
From: Ashar Siddiqui [mailto:siddas...@gmail.com]
Sent: 0
Sean,
Do srst auto-prov none and then just create ephones (as many as required)
and put the following in there:
Ephone 1
No privacy
!
Ephone 2
No privacy
!
You will need to do all the basic requirements for Cbarge like conference
hardware, sdspfarm units etc and configuring dspfar
Sorry Folks not providing details in first attempt.
Thanks for all and special thanks to Matthew Berry and Randall Saborio for
their interest and figured out this issue.
Let’s make thing more understandable.
I am working in my home lab. I am trying to spoof call manager. My target is
to get musi
hi,
doesn't this break the 2nd requirement? I've never tried this, but I would
configure ephone-template and assign it to srst via "srst ephone template"
command of telephony-services.
sccp group should indeed be configured with srst as the 3rd option.
regards
kobel
On Fri, Jul 2, 2010 at 4:29
It will not.
You are not configuring ephones, you are just configuring privacy thing.
Ash>
From: kobel [mailto:findko...@gmail.com]
Sent: 02 July 2010 16:00
To: Ashar Siddiqui
Cc: sean hurricane; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] SRST
hi,
doesn't this bre
Ashar,
I have tried your configuration and it does not satisfy the
requirementi will try Kobel's configuration later..
On Fri, Jul 2, 2010 at 11:33 AM, Ashar Siddiqui wrote:
> It will not.
>
> You are not configuring ephones, you are just configuring privacy thing.
>
>
>
> Ash>
>
>
Dear All,
I face problem with br1 router it seems register as in fallback mode
(I add the br1 router in cucm) and did the configuration as attachment , and
no mgcp then mgcp also I restart the router (also I removed ccm-manager
fallback-mgcp )
I can register hq-router and br2 router as mgcp withou
d to use
auto-provison none but then cbarge does not work.
any thoughts?
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What is the best way to accomplish this without using srst auto-provision
all which clearly does not meet the requirementI have tried to use
auto-provison none but then cbarge does not work.
any thoughts?
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please show us "sh ephone sum" and a show run. both from teh branch
router, and tell us what interface the phone is on.
-Peter
On Fri, Jul 2, 2010 at 7:52 AM, Afzal Bhutta wrote:
> Sorry Folks not providing details in first attempt.
>
> Thanks for all and special thanks to Matthew Berry and Rand
2
>
> No privacy
>
> !
>
>
>
> You will need to do all the basic requirements for Cbarge like conference
> hardware, sdspfarm units etc and configuring dspfarm with
telephony-service
> address on third priority if the requirement is that Cbarge is working
> duri
Is your Site B router MGCP or H323? With an H323 gateway I could get the
router to stream the local piano music while the MoH server is set to one hop
in UCM. With an MGCP gateway I couldn't get this to work and it always streams
from UCM unless the router is in SRST mode then it plays piano mu
ed)
> > and put the following in there:
> >
> >
> >
> > Ephone 1
> >
> > No privacy
> >
> > !
> >
> > Ephone 2
> >
> > No privacy
> >
> > !
> >
> >
> >
> > You will need to do all the basic requirement
Hey Voice Team,
We recently made some design changes to our Voice bootcamp offerings to
extend our PL infrastructure directly to the bootcamp phones. Meaning The
Voice candidates in our bootcamps are able to test QoS, SRST, and other
services offered at each branch site using the phones in the
Hi,
I face a problem over a sip trunk, add transformation pattern calling
and calles css under Outbound Calls (I uncheck use device pool.tcss..)
When I try to do tranformation for called and calling party under the rp it
worked fine.
please advice
__
I think its easy to test with a simple acl on hq-rtr.
If br1phn2 is 192.168.5.13, my acl is
ip access-list extended SRST
deny ip 10.10.201.0 0.0.0.255 10.10.210.0 0.0.0.255
deny ip host 10.10.110.2 10.10.210.0 0.0.0.255
deny ip host 192.168.5.13 10.10.210.0 0.0.0.255
permit ip any any
Appli
Brian,
What I am speaking of will make the phones appear as if they are directly
connected to the devices in the rack. In an optimal setup they will show up as
CDP neighbors even on BR1, BR2, and HQ-3750. I am not sure you are
understanding what I am proposing.
Regards,
Tyson Scott
Hello,
I am having trouble with calls coming into UCM when using SIP early offer. I
have inbound fast start selected on my trunk. Outbound calls from HQ and BR1 to
BR2 work fine. When I call 5002 from 3002, I get a fast busy.
If I look at the voip ccapi inout debugs on HQ-RTR when I try callin
Hi Ashar,
thks, yes what I mean is ccie lab.
When I use 2nd option, it is specific ,agree.
However ,I can not capture , let say :
1. 3432143...
2. also when I need it for TEHO
I have to create other specific one.
If 1st one ,no drawback in ccie lab , it save some time for me in exam :)
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Message: 2
Date: Sat, 3 Jul 2010 00:27:21 +0300
From: amr thabt
To: ccie_voice@onlinestudylist.co
t.com
<http://www.ipexpert.com/>
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Date: Sat, 3 Jul 2010 0
For HQ, streaming from CM, it would make sense to increase the Max Hops.
The way I understand it, the max hops would account for any extra
interface that the stream has to go through. So for example, from CM
towards default gateway on servers VLAN is 1 hop, then towards voice
vlan default gateway
Hi,
Pls send the running config from both the routers.
Things to check:
1. Since you are getting unallocated number first thing to check is the
outgoing sip dial-peer (with session protocol sipv2) pointing to the
call-manager)
2. do u have the following commands under voice service voip on t
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