What are you looking at in the debugs and which leg is this ?
you said you have CUBE in between so you will see 2 seprated H245
negotiation for each leg ,
can you post the H225 and h245 debugs for me please ?
just to make sure that we both talking about the same thing , this
call is Slow start a
They are real routers.
Cisco IOU can't fake voice interfaces or DSPs.
As per conversation at CiscoLive.
The phones are connected over L2 VPN to be able to keep all lab equipment
remote from test centers.
Rumor is that RTP and SJ are now homed in Texas.
On Thu, Dec 1, 2011 at 10:21 PM, Ken Wyan
It's real router of course.
In actual lab, my router even crashed before
Sent from my iPhone
Pls pardon my fat fingers.
On 2 Dec, 2011, at 11:21 AM, Ken Wyan wrote:
> Hi,
>
> Does CCIE Voice lab routers are real or run on IOU?
>
> Guys say not to restart IOU during lab
>
> _
Hi All,
As per your suggestion, I bind SIP to voice vlan ip address & stopped the CM
service on both the CUCM servers & it started to work perfectly.
Thank you all for your help :-)
T&R
From: Ashraf Ayyash
To: Gurpreet Singh Kukreja
Cc: Raees Shaikh ; "cci
Hi All,
I have my own pod at home. After performing VOD 5.5 (Gatekeeper Remote Zone), I
still could not call India PSTN 916745738932 from HQ phone 2 (getting reorder).
Here is the config of HQ and PSTN-WAN. I am sending only 916745738932 to PSTN
router as shown i the debug gatekeeper main 10 be
I noticed a weird thing while testing MGCP.
If I call out to my pstn phone and answer the call by pressing the answer
soft key , the call will disconnect after about two minutes.
If I answer the call with the speaker button , it stays up forever.
I'm guessing the problem is I don't have handsets
Hi,
Does CCIE Voice lab routers are real or run on IOU?
Guys say not to restart IOU during lab
___
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com
Are you a CCNP or CCIE and looking for a job? Check out
A what ?
On Thu, Dec 1, 2011 at 9:10 PM, Ken Wyan wrote:
> Did you try adding a ephone-dn (octo line) with conference adhoc for
> cBarge?
>
> On Fri, Dec 2, 2011 at 5:24 AM, ccielabrat wrote:
>
>> Hey all,
>>
>> I'm finishing up testing my CME as SRST testing and ran in to a problem
>> where i
Did you try adding a ephone-dn (octo line) with conference adhoc for cBarge?
On Fri, Dec 2, 2011 at 5:24 AM, ccielabrat wrote:
> Hey all,
>
> I'm finishing up testing my CME as SRST testing and ran in to a problem
> where it looks like the phones are getting the ephone-dn-template
> configured u
Hi Ashraf,
I removed everything but the one set of files from the directory and
ran "Software install add .pkg
but it didn't change the behavior.
On Thu, Dec 1, 2011 at 3:16 PM, wrote:
> oh, ok.
> I'll give it a try.
>
>
> On Thu, Dec 1, 2011 at 1:44 PM, Ashraf Ayyash
> wrote:
> > Hello ,
> >
Hey all,
I'm finishing up testing my CME as SRST testing and ran in to a problem
where it looks like the phones are getting the ephone-dn-template
configured under telephony-service.
The phone just rings and rings
I'm also wondering if the DND button should automatically push calls to
voicema
ok, I'm getting to understand this better.
I don't see any mention of a tcs failure though
See the output of debug h245 asn1 below. Where is the indication of a
failure?
Also, I have CUBE running with a Hw transcoder registered locally on HQ
telephony-service.
I would think the CUBE should alloca
oh, ok.
I'll give it a try.
On Thu, Dec 1, 2011 at 1:44 PM, Ashraf Ayyash wrote:
> Hello ,
>
> why you have tow packages in the root directory ?
>
> you have to have the full package of 7.0.6 and the lang pack of GB
> 7.0.6 ONLY on the root directory , run the installation again and see
> how it
The ccapi debug will show you the cause code which doesn't explain why
the call failed ,
you have to debug the h245 asn1 and check the TCS and see the codecs
advertised and received and then you will get the TCS negotiation
failure so you can explain that there is codec mismatch
Ash
On Thu, Dec
Thanks Ash - Yeah I'll speak to BT thought this was going to be a tough one.
-Original Message-
From: Ashraf Ayyash [mailto:ash.ayy...@gmail.com]
Sent: Thursday, December 01, 2011 11:39 AM
To: Sears, Michael (msears)
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] [EMEA
Hello ,
why you have tow packages in the root directory ?
you have to have the full package of 7.0.6 and the lang pack of GB
7.0.6 ONLY on the root directory , run the installation again and see
how it will go
Ash
On Thu, Dec 1, 2011 at 8:02 AM, ccielabrat wrote:
> Hi Ashraf,
>
> See below. Th
Germany Dial-plan is the most complicated dial-plan in the world
because its not organized as the NANP or UK Dial plans , they don't
have a fixed pattern in the access codes or length , you have to
speak to someone from Germany who know about the telecom there or
maybe the German telecom itself ca
dont debug the code level, deb h245 asn1 will show you the tcs
negotiation failure
Ash
On Thu, Dec 1, 2011 at 8:12 AM, ccielabrat wrote:
> I'm trying to setup a call from HQ CUCM via GK-Trunk to a Remote Gk Zone.
>
> I have the Gatekeeper configured with OutVia for the remote zone referencing
>
Great Post. Thanks you so much.
I will test it in my LAB tomorrow and will post the results as well.
Initially my issue was caused by the incorrect dspfarm profile for Xcoder.
I missed to put the G729r8 codec over there, so the Xcoder did not work,
but as soon as I have added that codec to Xcoder
Use the command debug voice ccapi inout. H323 debugs won't show in this case.
Regards,
Mohammed Al Baqari
Sent from my iPhone
On Dec 1, 2011, at 6:12 PM, ccielabrat wrote:
> I'm trying to setup a call from HQ CUCM via GK-Trunk to a Remote Gk Zone.
>
> I have the Gatekeeper configured with O
It is a Cisco provided solution to get around issue with multiple VXML
transfers for the same call. CUCM version 7.1.3 and H323 gateway on IOS
15.0.1(M)
We found that when a call needed to be transferred (not CUCM xfers) within
VXML app more than twice it would not complete and disconnect. Introdu
This is an automated reply to your message "CCIE_Voice Digest, Vol 70, Issue 4"
sent to stewart.mcfarl...@provista-uk.com.
Dear ccie_voice-requ...@onlinestudylist.com
Thank you for your email.
Please note that due to unforseen circumstances I will be out of the office for
the rest of this we
I have tested this and got the result successful using MTP + XCODE. I used
exactly same scenario. Here is the output.
HQ#sh call leg act su
G L Elog A/O FAX T Codec typePeer Address
IP R:
G0 L 83 N ORG T10g729r8 VOIPP
142.2.66.254:17734
Priyank,
Question is WHY MTP?
On Thu, Dec 1, 2011 at 4:33 AM, Priyank Kiran wrote:
> Thanks, now do you guys know if this is going to be addressed in later
> revisions of CUCM (8.x) OR changing from H323 to SIP would help?
>
> OR may be its just the way MTP behaves and you just can't support MMo
I'm trying to setup a call from HQ CUCM via GK-Trunk to a Remote Gk Zone.
I have the Gatekeeper configured with OutVia for the remote zone
referencing a CUBE on the HQ router.
I didn't realize (but it makes sense now) that with "Wait for H.245"
unchecked on on the CUCM trunk, the call setup goes
Hi Ashraf,
See below. Thank you!
ftp> ls
200 Port command successful
150 Opening data channel for directory list.
cue-installer.nm-aim.7.0.1
cue-installer.nm-aim.7.0.6
cue-vm-en_GB-langpack.nm-aim.7.0.6.prt1
cue-vm-full-k9.nm-aim.7.0.1.prt1
cue-vm-full-k9.nm-aim.7.0.6.prt1
cue-vm-installer-k9.nm-
Greetings - I am seeking input on developing a dial plan for a site that has
been thrown my way in Munich Germany. I'm new to ISDN ERA and have been using
NANP for years. Any input regarding developing a dial plan for Munich Germany,
including sample configurations of CUCM and Gateways, would
When you have Voice Class Codec for incoming Dial-peer on CUCME, with G711
and G729, the calls to CUE are not transcoded, and the calls are failing.
That's because CUE uses only G711 and as the G711 is also found in the
Voice Class codec, the call is trying to take G711, whereas CUCM IP Phones
are
Tks Ash!
That explains the solution guide where there are no voice classes on thIs trunk,
although in trunks to PSTN breakout Vik invariably puts the voice-class in the
dial peer.
On 1 Dec 2011, at 15:41, Ashraf Ayyash wrote:
> Hello Anthony ,
>
> You cannot Transcode call that Hit Dial pee
Hi,
I have the required devices to study CCIE LAB v.3, it does not match the
requirements exactly but it is very good. To save my time I need to upload the
initial configuration for each device at the beginning of each lab.
I need to build an automated way to do this rather than do everything
Hi,
If I want to add one softkey to a certain state of a phone , I have to
create a ephone template with all the required softkeys for the state &
apply to ephone.
For that I have to check other default softkeys by making a call from this
phone & checking softkeys. But this is time consuming.
I
This mean you are installing the Wrong Language files , or you missing
on critical file ,
can you please paste what you have in the FTP directory root ?
Ash
On Wed, Nov 30, 2011 at 1:03 PM, ccielabrat wrote:
> I'm trying to add a second language to an AIM-CUE.
>
> I use the command "software in
The show ccm music will not show you any output for Voip calls , this
is used when you call from the PSTN to the BR1 and place the call on
hold ( in other word when the DSP get involved as the ccm music
command is to make the DSP injecting the MOH stream into the Isdn Call
) so this is normal ,
I
33 matches
Mail list logo
