Guys,
I passed the lab yesterday on second attempt. I appreciate all
the help I have gotten from OSL.
Regards,
Emin
___
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www.ipexpert.com
Are you a CCNP
Looks like it is failing due to codec negotiation.
Media negotiation failure
Typical scenarios include:
*No codec match occurred.
*H.323 or H.245 problem leading to failure in media negotiation
65
CC_CAUSE_BEARER_CAPABILITY_
NOT_IMPLEMENTED
Indicates that the equipment sending this cause does
Three different configurations are available for the switch interface. The
configurations are bandwidth shape, share, and limit. You can also configure
egress queue 1 as the priority queue. If the priority queue is enabled, SRR
services it until it is empty before servicing the other three
Adam,
I have had the same exact question. As far as I know, you could
only assign bandwidth to inbound priority queue. If someone sheds a light on
this, it would be appreciated.
Thank,
Emin
From: ccie_voice-boun...@onlinestudylist.com
Guys,
Do you manually add max-reserved-bandwidth 100 statement
under the qos applied interface? Looks like it is not part of the auto-qos.
Only affects the qos allocations when percent% is used.
Thanks,
___
For more information
Guys,
CUBE on GK setup scenario. I have media flow-through defined
under the CUBE dial-peer. CUBE h323 is bound to Voice Vlan. Whenever the voice
stream is established, is see that outbound voip leg(sh voip rtp conncections)
is sourced from Server Vlan/SVI ip address instead of
Rogers,
On branch sites, since there is no LAN QoS, frames/packets
arrive in with proper L3 markings. My practice has been to trust the markings
on the branch sites. Any particular reason why you don't do it on the branch
sites?
Thanks,
Emin
From:
Sidenote: NTP sync on pub will templorarliy bring down the replication to 3
on pub( I have timed it to 10 minutes). In that case you just have to wait.
From: Ki Wi [mailto:kiwi.vo...@gmail.com]
Sent: Sunday, June 05, 2011 3:10 PM
To: Emin Guliyev
Cc: ccie_voice@onlinestudylist.com
Subject: Re
Congrats!!!
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Rogers Ochieng
Sent: Friday, May 20, 2011 2:38 AM
To: ccie_voice@onlinestudylist.com
Subject: [NEWSENDER] - [OSL | CCIE_Voice] Passed CCIE#28970!!! - Message is
from an
Hey guys,
I will be going for my 1st attempt this month. Wanted to verify
some minor details. Are we supposed to configure correct cadence/locale on
endpoints and gateways? Also, do you lose any points for mismatching timezones
on UC mailboxes? For afterhours mailbox, are we
Hey guys,
Just wanted to clear this up. So, If I use mls qos trust cos
on an interface, from then on all the queue map, threshold map will be based on
Cos-Dscp table? That is also the same for mls qos trust dscp dscp-cos tables
will be used to convert layer2 markings to layer3
To: Emin Guliyev
Cc: Shrini; Bill Lake; OSL Questions
Subject: Re: [OSL | CCIE_Voice] SIP early media on CME
and try one thing, I don't like having a sip dial-peer (session protocol sip)
on incoming RAS call...
I've never tried that, but something can go wrong... to start with you have
dtmf-relay
Congrats! Now that a job well done :)
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Shrini
Sent: Wednesday, May 04, 2011 4:02 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Finally succeeded ..Got CCIE
Hi Experts,
Guys,
Quick question: Why is it that even though I explicitly define
codec g711ulaw under voice register pool. When the INVITE is sent out SDP
always includes all the codecs and DTMF types. It is also the same for DTMF
relay. Even though I put in rte-nte only, it offers
PM
To: Emin Guliyev
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] SIP early media on CME
Hi Emin,
I've noticed this also. My bet would be that voice register global
configuration affects incoming dial-peers created internally by CUCME. Even if
IP phone supports all codecs
invoked.
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
From: George Goglidze [mailto:gogli...@gmail.com]
Sent: Tuesday, May 03, 2011 5:50 PM
To: Emin Guliyev
Cc: ccie_voice
no vad
Thanks,
Emin
-Original Message-
From: Bill Lake [mailto:whl...@gmail.com]
Sent: Tuesday, May 03, 2011 7:21 PM
To: Emin Guliyev
Cc: OSL Questions
Subject: Re: [OSL | CCIE_Voice] SIP early media on CME
It seems that you have a dial-peer with voice-class codec
Cc: Emin Guliyev; OSL Questions
Subject: Re: [OSL | CCIE_Voice] SIP early media on CME
By default voip dial-peer sends g729. To use G711 you have to define codec in
outgoing dial-peer.
sample config :
dial-peer v 1 voip
desttination-pattern 1...$
codec g711ulaw
session-target ipv4:X.X.X.X
Just wanted to share this with you guys:
What is the difference between g729r8, g729ar8, g729br8, and g729abr8 ?
The format of the codewords generated by the above 4 codecs are identical.
The g729ar8 is a reduced complexity version of the G.729 speech codec.
It is bit stream interoperable with
If you are specifically looking for codec negotiation run debug h245 asn1 /
events
Thanks,
Emin
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