Assuming your cm service is started and your phones have ip connectivity to
your subscriber, it sounds like a replication issue to me. Happened to me when
I booted my vmware servers the wrong way - replication would always break and
nothing would ever register with my sub. Hope this helps.
SecureCRT was mentioned during my IPX bootcamp, so it should be public
knowledge. As for version information, that I have no idea about.
-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Brian Valentine
Sent:
I'm still around.. just in stealth mode :)
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vik Malhi
Sent: Wednesday, November 11, 2009 12:13 AM
To: Aamir Panjwani; Kumar, Narinder; Robert McGhee; OSL Group
Subject: Re: [OSL | CCIE_Voice]
...@onlinestudylist.commailto: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#25711 (Daniel Rodriguez)
--
Message
Have you updated the domain name under SIP Proxy service parameters?
- Dan
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Omotayo
Sent: Tuesday, November 03, 2009 11:14 AM
To: OSL Group
Subject: [OSL | CCIE_Voice] Display on the CUPC
What's the status of CUPC after you log in? If Available is greyed out, I
would check out the diagnostics report for any configurations that may be
missing.
- Dan
From: Omotayo [mailto:adefilabi...@gmail.com]
Sent: Tuesday, November 03, 2009 11:27 AM
To: Daniel Rodriguez
Cc: OSL Group
Subject
[mailto:adefilabi...@gmail.com]
Sent: Tuesday, November 03, 2009 12:47 PM
To: Daniel Rodriguez
Cc: OSL Group
Subject: Re: [OSL | CCIE_Voice] Display on the CUPC phone
Could it be the version am running?
On Tue, Nov 3, 2009 at 5:44 PM, Omotayo
adefilabi...@gmail.commailto:adefilabi...@gmail.com wrote
I completely agree with Phil on his comment about strategy. Strategy is key -
if you have a strategy, it's very important to stick with that strategy and
stay with what you're comfortable with. Like Phil, I disliked the idea of
grouping tasks by technology. I always thought that type of
Folks:
Just wanted to thank a few people here for helping me with some of my questions
over the past few weeks... and a huge thanks to IPExpert for the very helpful
bootcamp and practice lab material.
- Dan
___
For more information regarding industry
on the other side
and flipped through them whenever possible.
- Dan
From: Wilson Bolanos [mailto:wbola...@pvt.com]
Sent: Friday, October 30, 2009 1:14 PM
To: Daniel Rodriguez; A A; ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] [NEWSENDER] - RE: CCIE #25689 - Message is from
Hey Jeff - It sounds like you're failing over from a SIP trunk to an MGCP
gateway, but I read that you're shutting a voice-port. Did you mean H323
gateway instead of SIP Trunk? Sorry but I don't have the lab manual in front of
me. If you meant H323 gateway to MGCP, make sure your service
You're correct - its from the perspective of the egress gateway. For example,
using IPExpert labs as a point of reference, calls from HQ Gw to Spain would be
international. That is, you pass the international access code and country code
to the PSTN with the called number type as international.
requirements, but
good to hear you got it working.
- Original Message -
From: Girard, Jeffrey COL MIL USA jeffrey.gir...@us.army.mil
To: Daniel Rodriguez; ccie_voice@onlinestudylist.com
ccie_voice@onlinestudylist.com
Sent: Fri Oct 30 22:02:27 2009
Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 5.8
I would say it depends on what I'm being asked to do. If nothing is specified
in terms of MGCP failover/switchback behavior, nothing - I would leave it at
default. If I'm being asked to configure MGCP failover behavior so that my MGCP
gateways don't switchback until I'm confident my server or
...@ivision.com.au
To: Daniel Rodriguez
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Sent: Fri Oct 30 22:43:13 2009
Subject: RE: [OSL | CCIE_Voice] mgcp switchover: graceful or immediate
Thanks Dan...what about h323 call preservation? When there is a active call
through h323
Are you using an H323 gateway? I don't believe + is supported in H225 call
setup. You'll need to prefix it using a translation rule.
- Original Message -
From: ccie_voice-boun...@onlinestudylist.com
ccie_voice-boun...@onlinestudylist.com
To: OSL Group ccie_voice@onlinestudylist.com
The difference you see is due to the requirements of the connected hardware.
For your 3750 at HQ, you configure the switchport as an access port. When
connected to EtherSwitch (such as your BR1 and BR2 routers), you configure your
switchport as a trunk. Although its configured as an access port
Are these calls going to SIP phones at BR2? I don't recall if this was asked
before so disregard if I'm being redundant, but have you eliminated codec
negotiation as the cause by assigning your dial-peers and or regions for g711u
across all?
Dan
- Original Message -
From:
Hmm. On your H323 gatewat configuration page do you have Wait for far end
Terminal Capabilities Set checked off..? If so, try unchecking and test again.
Dan
- Original Message -
From: Omotayo adefilabi...@gmail.com
To: Daniel Rodriguez
Cc: OSL Group ccie_voice@onlinestudylist.com
Sent
I don't believe UC demo licenses include VPIM capabilities. I've run into this
in the past as well on my VMWare server running a demo license.
- Dan
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Daryl Smith
Sent: Monday, October 26,
You should only need the AXL user if you plan on importing your users from
CUCM. If you're importing users, you'll need to create your AXL user in CUCM,
define it in UC, and associate a new AXL server with your phone system in UC.
Otherwise you can manually create users in UC, completely
Hey CT..
Can you send a show run output?
- Dan
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ctpresident
Sent: Saturday, October 17, 2009 7:22 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] voice_port is not
[mailto:j.jho...@gmail.com]
Sent: Saturday, October 17, 2009 9:49 AM
To: Daniel Rodriguez
Cc: kevin.dami...@vitalsite.com; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] trasnfer to VM
What I am tring to do is make the hunt pilot use the transfer to VM. So In my
case if a number
assigned (which
will pass 1200 to voicemail).
Hope that helps!
- Dan
From: J Hogan [mailto:j.jho...@gmail.com]
Sent: Saturday, October 17, 2009 11:04 AM
To: Daniel Rodriguez
Cc: kevin.dami...@vitalsite.com; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] trasnfer to VM
Call commed
...@gmail.com]
Sent: Saturday, October 17, 2009 12:11 PM
To: Daniel Rodriguez
Cc: kevin.dami...@vitalsite.com; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] trasnfer to VM
Thanks
I leaft everything the way it was except I took your advice and added the
alternat Extensin and it works
You have to download the CME file package off CCO. Also make sure to download
the files for the CME version you're running on the 2801.
- Dan
From: ccie_voice-boun...@onlinestudylist.com
[ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Cotter
If this is related to the IP expert labs, I recall the PSTN configs only
allocating 3 B-channels for each PRI.
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of P N
Sent: Friday, October 16, 2009 12:52 AM
To: ccie_voice@onlinestudylist.com
The CUCME book provides detailed information and call-flow examples related to
H323 ECS and MTP.
Check out chapter eight... The title of the book is Cisco IP Communications
Express (CME with CUE). The CM Fundamentals book also provides info on this
topic, but the CME book is much better at
The only single reference guide out there I know of is published by Cisco. It's
an e-book called CCIE Quick Reference Sheets. I found it somewhat useful, but
its far from any other all-in-one CCIE book released by Cisco. I used it to get
an understanding of what was on the exam and went to the
Few things... Are your RSVP agents registered to CUCM? Are they available for
media resource allocation at all sites (in your custom MRGLs or the null
group)? Do you have ip rsvp bandwidth statements under your IP interfaces along
the IP path? Hope that helps.
- Dan
- Original Message
should see voice message indication in CUPC. Unfortunately I can't help
with dialing into VM by clicking the voice message notification in CUPC.
Hope that helps.
- Dan
From: Omotayo [mailto:adefilabi...@gmail.com]
Sent: Wednesday, October 14, 2009 10:45 AM
To: Daniel Rodriguez
Cc: Vik Malhi
Is this regarding SIP IP phones? If so, I believe this task was meant mainly
for the phone models that don't support KPML. The older models required you to
either send the digits en bloc. You would create dial plan patterns and push
them to the IP phone. It's great for emergency services - your
Hmm.. Assuming they mean transfer to a live operator/receptionist, the operator
input wouldn't necessarily require its own CH for the transfer.
It's possible that caller input 0 is configured for Attempt Transfer to an
existing CH within the 7 mentioned below, where as the other caller input
Is your translation pattern used for inter-office dialing (ex HQ to BR1)?
If so, I would work around this by making your xlate pattern more specific,
like 100X, 500X, and 300X. You wont overlap your outbound dial plan or anything
else as more configs are added. That's how I setup my inter
Folks:
I'm facing an issue with calling party transforms on my gateway
that I can't seem to figure out. It's related to lab 3 section regarding calls
between sites during WAN failure (ex. HQ call BR1 when BR1 in SRST).
Here's my configuration... I'll stick to my example between
and num).
Dan
From: Michael Ciarfello [mailto:mciarfe...@iplogic.com]
Sent: Friday, October 09, 2009 5:04 PM
To: Daniel Rodriguez; ccie_voice@onlinestudylist.com
Subject: [NEWSENDER] - RE: [OSL | CCIE_Voice] Calling XForm on GW - No
Transformation Being Applied - Message is from an unknown
for the tip.
- Dan
From: Kevin Damisch [mailto:kevin.dami...@vitalsite.com]
Sent: Friday, October 09, 2009 4:56 PM
To: Daniel Rodriguez; ccie_voice@onlinestudylist.com
Subject: [NEWSENDER] - RE: [OSL | CCIE_Voice] Calling XForm on GW - No
Transformation Being Applied - Message is from an unknown sender
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