How To Configure Time Synchronization for Cisco CallManager and Cisco Unity
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a008009470f.shtml
Date: Wed, 27 Aug 2008 23:48:26 -0700From: [EMAIL PROTECTED]: [EMAIL
PROTECTED]: [OSL | CCIE_Voice] How to synchronize NTP
Should be 7200 for a g729 requirement.
Vad is bad for fax
- Mensagem Original -
De: Kapil Atrish [EMAIL PROTECTED]
Enviada: domingo, 21 de setembro de 2008 23:49
Para: ccie_voice@onlinestudylist.com
Assunto: [OSL | CCIE_Voice] Fax question
Hi,
When Fax-Relay is enabled on 6608
Hi all,
Supose that we have to create a web custom administrator for CCME that has a
subset of the system admin.
How could we get the xml.template from the CCME flash to edit it?
I've tried with copy flash: tftp: to CCM server to edit there, but w/ no
success.
Regards, Sergio.
Jacob,
I don't know if it is related, but your dial-by-extension (3) is conflicting
with your ephone-hunt w/ pilot 4300. Right?
Sergio.
Date: Sat, 27 Sep 2008 16:26:00 -0400From: [EMAIL PROTECTED]: [EMAIL
PROTECTED]: Re: [OSL | CCIE_Voice] BACD
With the configuration below would anyone
Test, thank you.
_
Cansado de espaço para só 50 fotos? Conheça o Spaces, o site de relacionamentos
com até 6,000 fotos!
http://www.amigosdomessenger.com.br
Cme or cm?
- Mensagem Original -
De: Rachell Thornton [EMAIL PROTECTED]
Enviada: sexta-feira, 17 de outubro de 2008 21:44
Para: ccie_voice@onlinestudylist.com
Assunto: [OSL | CCIE_Voice] Fast Busy
I am trying to get my dial plan to work but my phone keeps giving me a fast
busy when I
Omar,
This document suggest some actions.
Understanding FXO Disconnect Problem
http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00800ae2d1.shtml
Date: Sat, 18 Oct 2008 01:46:30 -0700From: [EMAIL PROTECTED]: [EMAIL
PROTECTED]: [OSL | CCIE_Voice] PSTN to PSTN transfer
Jeremy,
Check if your MC3810 requires no use-proxy FXSZONE default inbound-to
terminal under gatekeeper configuration.
It worked for me with a ATA (h323) for calls *from* CallManager.
HTH
Sergio.
Date: Sat, 18 Oct 2008 14:50:03 +1100From: [EMAIL PROTECTED]: [EMAIL
PROTECTED]: [EMAIL
Steve,
You can try as wrote by Mark this last week:
You can 'hide' the internal number from other phones by placing it in a
isolated Partition, and then create a Xlate pattern that matches the internal
number you wish to call- and place it in a Partition that can be reached from
other
ops, you have to upload the Prompt w/ Prompt Management Menu.
Sergio. From: [EMAIL PROTECTED] To: [EMAIL PROTECTED];
ccie_voice@onlinestudylist.com Date: Tue, 11 Nov 2008 02:25:53 + Subject:
Re: [OSL | CCIE_Voice] IPCC custom prompt James You have to upload the
script with Script
Steve,
You are right. Vik wrote this some time ago:
The QOS markings on Unity are really dependent on the version of the TSP.
With Unity 4.0(5) the TSP (8.0x) hardcodes the DSCP markings to AF31 forSCCP
and EF for Media. With Unity 5.x and TSP 8.2x you can change themarkings in
UTIM
Hi,
My CAD can not logon. The system says The ID you entered was not found
I have checked at CM the ID, password, CTI check box and the rmjtapi
associtation to the device. But I can not find what is wrong.
The AA.aef default script works fine. And a reboot does not fix the problem.
May
Narinder, thanks for your reply. The DB was corrupted.
During the IPCC installation, the system says to do not restart CRS Engine
at any node for 10 minutes.
But, after few minutes I've alredy configured the jtapi and I did restart
the CRS. So...
Cheers, Sergio.
show up in IPCC as a resource?
On Tue, Dec 2, 2008 at 8:55 AM, Sergio Polizer [EMAIL PROTECTED] wrote:
Hi,
My CAD can not logon. The system says The ID you entered was not found
I have checked at CM the ID, password, CTI check box and the rmjtapi
associtation to the device. But I can
Hi,
The Q.22.39 asks for mark, at Cat3550, DSCP PHB label as EF for RTP traffic
from the port connected to IP Phones.
Proctor Guide states for a MQC config. So, I'd like to know if just a map
cos-dscp also answered the question. For example:
mls qos
mls qos map cos-dscp 0 8 16 24 32 46 48
PHB label as EF for RTP
traffic from the port connected to the:
- CM servers, I'd use the mls qos trust dscp.
- gateway, I don't see a way unless by MQC.
Cheers, Sergio Polizer.
Date: Wed, 17 Dec 2008 18:23:35 -0600
From: ryanstudyvo...@gmail.com
To: spoli...@hotmail.com
Subject: Re: [OSL
Narinder,
Check with perfmon if the xconder has been involked. In a negative case, see
the CTI MRGL.
Sergio.
From: narinder.ku...@uxcg.com.auto: ccie_vo...@onlinestudylist.comdate: Sun, 21
Dec 2008 00:38:59 +1100Subject: [OSL | CCIE_Voice] IPCC CTIRP
I have a 6608 xcoder at SiteA
Anil,
I don't kwon if I understood the requirements but:
If you set a dn 3001 w/ dual-line and 3101 with call-forward busy to 3001 Will
It answer the question?
By default, the CME line has call-waiting beeps enable.
Sergio.
Date: Fri, 26 Dec 2008 21:02:11 -0800
From: anil...@yahoo.com
To:
I have seen in my labs that after change de bandwidth total default I had to
shut and no shut the gatekeeper to the change start to work fine.
Sergio.
Date: Sun, 28 Dec 2008 19:08:35 -0800From: anil...@yahoo.comto:
ryanstudyvo...@gmail.comcc: ccie_vo...@onlinestudylist.comsubject: Re: [OSL
Jeremy,
Check if you have configured:
the voice card to support transcoding and
conferencing.Gateway(config-voicecard)#dsp services dspfarm
and enable DSP farming.Gateway(config)#dspfarm Sergio.
Date: Mon, 29 Dec 2008 13:19:46 +1100From: jeremy.coo...@gmail.comto:
it works!
Congrats, Alex.
Date: Mon, 29 Dec 2008 09:18:41 -0800From: anil...@yahoo.comto:
narinder.ku...@uxcg.com.au; ccie_voice@onlinestudylist.com;
alex.arsen...@gmail.comsubject: Re: [OSL | CCIE_Voice] Forwarding a PSTN
incoing call to extension's VM on CME
Thanks a lot Alex !!!
This post has some days but is similar that we are talking about transfers to
CUE.
Someone could tested this config proposed bellow? For me the incoming call stay
ringing at 3001.
But we can have One button VM transfer at CME phone to CUE, configuring
call-forward noan VM nº timeout x and
Hi,
During SRST, calls transferred to Voice Mail works fine if the SRST gw and PSTN
could send the RDNIS property. There are some workarounds when this is not the
case.
At Vol 3-Lab 2, Proctor Guide states for ex. 38 that when a call is transferred
to voicemail by BR1 phone, It arrives
Thanks for your reply, Jianhong.
I checked the SRND and CFNR (call-forward unregistered) is available after 4.2.
So, the integration VM/SRST worked
for me IF in additon the config stated by Vol 3-Lab 2, Proctor Guide ex. 38, I
also transfer BR1 Phones:
-forward-no answer internal to 1 617
Also you may check if IGMP is enable at the 6500 or apply set igmp enable.
Sergio.
Date: Mon, 5 Jan 2009 10:24:37 -0600From: ryanstudyvo...@gmail.comto:
marwa_ah...@seegypt.comcc: ccie_vo...@onlinestudylist.comsubject: Re: [OSL |
CCIE_Voice] music on hold
Technically no you will need
Ante,
Sorry, but I think is not possible. Try to explain you reason tho cert support
team.
http://ciscocert.custhelp.com/cgi-bin/ciscocert.cfg/php/enduser/std_alp.php
Regards, Sergio.
Date: Wed, 7 Jan 2009 10:10:28 +0100From: ante.bo...@combis.hrto:
I think that the CME will send a ARQ to the GK just IF the call to the BACD
match the outbound dial-peer voip with session target ras configured. Please,
correct me If I am wrong.
Regards, Sergio.
Date: Wed, 7 Jan 2009 11:01:49 -0600From: ryanstudyvo...@gmail.comto:
I'm seeing a similar problem, but at my case if I move the Device Pool of
software MTP from HQ to BR1 does not change the scenario. I don't know why.
I'm testing calls from CME to Hqph1 e Br1ph1. CME--SIP(g729)--- CM HqPh1 e
Br1Ph2
The regions are the classical configuration of g711
I think that in this case it will block the CLID for all calls that are going
out through this gw.
If you keep the GW with CLID=default (not allowed) is just fine and your config
at the RP to restrict the CLID will works.
Sergio.
From: narinder.ku...@uxcg.com.auto:
I lost my count how many times I have to restart the CM Service during a lab to
make some configs works.
E.g. Unity Integration, MGCP config changes, CTI ports, ICT Trunks and others.
I'd like to know if it is an issue from my VM's. Is the same for you?
I've really appreciate your
Hi Ryan and All,
I think the limitation is at the Site B. According to SRND, the router can
stream only a single audio file from flash and that you can use only a single
multicast address and port number per router.
If we had other MMoH sorce at Site B It will works.
Regards, Sergio.
According to Vik:
The DNIS-DIGITS command speeds up the setup time for inbound calls FROM
thePSTN. Given that proctorlabs are sending 10 digits all of the time min 10max
10 would be good too. This does not affect outbound calls
May someone kindly clarify the Layer 2 overhead of FR with MLP? Is 13 or 17
bytes?
Thanks in advance, Sergio.
Date: Tue, 10 Feb 2009 23:37:26 +0700From:
nattawut.boonpram...@atosorigin.comto: mo...@hotmail.com;
ccie_vo...@onlinestudylist.comsubject: Re: [OSL | CCIE_Voice] Compressed voice
Chris,
According to the QOS SRND bc = cir/100.
Sergio.
From: christopherc_56...@hotmail.comto: ccie_vo...@onlinestudylist.comdate:
Fri, 13 Feb 2009 11:14:03 -0600Subject: [OSL | CCIE_Voice] Frame-Relay
Traffic-Shaping Calculations
I'm trying to figure out the calculations for the
Scott,
Right. In the case of your ephone-hunt, maybe your Br2 pilot number was already
registered to the GK when you enter the no-reg at the ephone-hunt.
In this case, I only way that I know is reloading the Br2.
Sergio.
Date: Fri, 6 Mar 2009 01:30:57 -0500
From:
Can you try?
Voice service voip
h323
Call preserve
- Mensagem Original -
De: Balamurugan Singaram mmailb...@yahoo.com
Enviada: terça-feira, 10 de março de 2009 04:20
Para: cl...@mcglamry.net; scott.odonn...@gmail.com
Cc: ccie_voice@onlinestudylist.com
Assunto: Re: [OSL | CCIE_Voice]
If you apply the service policy input to the Interface Vlan of the Br1 could
answer the question?
From: kapilatr...@hotmail.com
To: vma...@ipexpert.com; ccie_voice@onlinestudylist.com
Date: Wed, 11 Mar 2009 01:24:01 +0530
Subject: Re: [OSL | CCIE_Voice] DSCP marking on NM-ESW module
I
Same here.
Ps. the PSTN also have isdn outgoing display-ie configured.
Any suggestions?
Thanks in advance, Sergio.
Date: Wed, 11 Mar 2009 22:26:06 +0100
From: christian.hennr...@intact-is.com
To: kapilatr...@hotmail.com
CC: ccie_voice@onlinestudylist.com
Subject: Re: [OSL |
Could be transfer-system pattern .T missing ?
- Mensagem Original -
De: marwa marwa_ah...@seegypt.com
Enviada: quinta-feira, 12 de março de 2009 08:53
Para: ccie_voice@onlinestudylist.com
Assunto: [OSL | CCIE_Voice] srst callforward all
hi,
i have site b in srst with h323 gw, i need if
. Look and see if you see a flag on the
redirect number that is 0xFF I'm betting that's the issue.
- Original Message -
From: Marwa Ahmed
To: Cliff McGlamry ; Sergio Polizer ; ccie_voice@onlinestudylist.com
Sent: Thursday, March 12, 2009 4:11 PM
Subject: RE: [OSL | CCIE_Voice
Your incomming dial-peer has to be a SuperSet cor list of the Outgoing dial
peer.
In your e.g. the corlist incoming css-pstn at dial-peer 1 does not have the
members of the corlist outgoing pt-911 .
Date: Thu, 12 Mar 2009 19:57:11 -0700
From: anil...@yahoo.com
To: ms...@ipexpert.com
CC:
I have tested and did not worked for me too. So I did some digits manipulations
with TP to get the same function.
I've seen some posts that are pointing for a feature that works with fxo only.
Date: Sat, 14 Mar 2009 13:17:09 -0700
From: e_er...@yahoo.com
To:
it should be ?
Thks
--- On Tue, 3/17/09, Sergio Polizer spoli...@hotmail.com wrote:
From: Sergio Polizer spoli...@hotmail.com
Subject: RE: [OSL | CCIE_Voice] SRST , default-destination not working
To: e_er...@yahoo.com, ccie_voice@onlinestudylist.com
Date: Tuesday, March 17, 2009, 6:45 AM
I have
Hi, I'm trying to make DTMF digits be recognize by Unity from CME calls through
a IPIPGW. This is the topology:
CME Phone ---h323--- IPIPGWSIPUnity
Please, May someone give me a suggestion? Thank You!
This is the config at CME:
dial-peer voice 21 voip
preference 1
from CME calls to Unity does not
work
To: ccie_voice@onlinestudylist.com; spoli...@hotmail.com
I believe it will not work since the conversion of dtmf digits from h245-a to
rtp-nte is not supported on Cisco SIP trunk side.
--- On Fri, 4/3/09, Sergio Polizer spoli...@hotmail.com wrote:
From: Sergio
an issue where you may be having a supplementary
services problem. If I'm right, you probably can't hold and unhold the
call either.
- Original Message -
From:
Sergio
Polizer
To: ccie_voice@onlinestudylist.com
Sent: Thursday, April 02, 2009 8:08
PM
Subject
, that's the
issue, please try following-
-Configure the dtmf-relay rtp-nte digit-drop command on
the incoming SIP dial-peer. On the H.323 side configure either dtmf-
relay h245-alphanumeric or dtmf-relay h245-signal command.
-anil
--- On Fri, 4/3/09, Sergio Polizer spoli...@hotmail.com
, that's the
issue, please try following-
-Configure the dtmf-relay rtp-nte digit-drop command on
the incoming SIP dial-peer. On the H.323 side configure either dtmf-
relay h245-alphanumeric or dtmf-relay h245-signal command.
-anil
--- On Fri, 4/3/09, Sergio Polizer spoli...@hotmail.com
this thread, could you
please pipe up? What was the allowed DTMF relay transformation and what
wasn't supported?
Cliff
- Original Message -
From:
Sergio
Polizer
To: cl...@mcglamry.net ; ccie_voice@onlinestudylist.com
Sent: Friday, April 03, 2009 1:49
PM
Subject: RE
Hi There,
I wonder what scenarios do you usually study with a FXS/FXO?
I can see the following:
1) Calls via CM SIP Trunk to FXS.
2) FXS/FXO controlled by MGCP.
3) FXS un/registered to a GK.
...
...
Thanks for your input.
Sergio.
I was taking a look at the sccp traffic generated by the router itself. E.g.
for harware transcoding.
And, I saw that the sccp ip precedence influence just the RTP pkts. The
default is 5.
All the sig pkts are not marked. So, It looks that we had to use the ip local
policy route-map.
Does
try enter login timeout under Telephony-service
and a restart all
Sergio.
Date: Thu, 16 Apr 2009 23:09:04 +1000
From: jeremy.coo...@gmail.com
To: cl...@mcglamry.net
CC: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CME call blocking question
Hi,
well, If Ialready logged in
Check if you have the following commands at sitec:
h323-gateway voip interface
h323-gateway voip id HQ-RTR ipaddr ip
h323-gateway voip h323-id BR2
h323-gateway voip tech-prefix 1#
h323-gateway voip bind srcaddr ip
use, show gateway.
Date: Fri, 17 Apr 2009 23:51:10 +1000
From:
Check if you have the following commands at sitec:
h323-gateway voip interface
h323-gateway voip id HQ-RTR ipaddr ip
h323-gateway voip h323-id BR2
h323-gateway voip tech-prefix 1#
h323-gateway voip bind srcaddr ip
use, show gateway.
Date: Fri, 17 Apr 2009 23:51:10 +1000
From:
HI,
I have a call from BR2 to HQ/BR1 in a ICT_GK with MTP checked to provide
supplementary services like hold/transfer.
My ICT have a BR2 DP that speak G729 with all others and a HW transcoding that
have HQ DP.
When I call from CME to HQ the call goes to HQ transcoder at G729 and connect a
need that, and don't have it in the MRGL, the call will
fail.
I'd also put the hardware resources ABOVE the software
resources. Then it should work with transfer and all.
- Original Message -
From:
Sergio
Polizer
To: ccie_voice@onlinestudylist.com
Sent: Thursday
Marwa,
I think that we have to both because:
- Policy-map marks the pkts that goes through the router
- With ip qos dscp cs3 signaling and mgcp ip qos dscp cs3 signaling marks
pkts thats are originated at router and are not inspect by the output
policy-maps.
Sergio
From:
Hi, Someone got make this to work?
Considere feature block offnet-to-offnet calls, but AAR calls can fw calls to
outside.
When HQ ph makes a call via AAR to SiteB, HQ ph can transfer to some line at
outside.
When SiteB ph makes a call via AAR to HQ,
Put both sw conf at the same mrg.
Sub first.
To test make 2 or 3 confs at the same time and you will see the load balacing.
Let us know how it does
- Mensagem Original -
De: Michael Ciarfello mciarfe...@iplogic.com
Enviada: sexta-feira, 8 de maio de 2009 22:22
Para:
Chris,
I was testing this and see that if you add the last and first name of the
remote users, is not necessary the work around via PDL.
remote username hqph2 fullname last pho
remote username hqph2 fullname first hq
cue show remote users detail
USERID
I just check and by this way CUE accepts the remote user if we enter with last
and first name.
If we were asked to enter the number directly We still have to use the PDL.
Regards, Sergio.
From: spoli...@hotmail.com
To: cpar...@cparker.us; ccie_voice@onlinestudylist.com
Subject: RE: [OSL
I did a search some time ago and i conclude that fxs controlled by cm is
supported at 4.2 or later
- Mensagem Original -
De: Onur Tufekci onurvc...@gmail.com
Enviada: domingo, 10 de maio de 2009 16:32
Para: ccieid...@gmail.com
Cc: ccie_voice@onlinestudylist.com
Assunto: Re: [OSL |
Try this
At the service parameter change those related to allow rerouting in
unallocated or busy number. Let us know how it does.
- Mensagem Original -
De: jeremy co jeremy.coo...@gmail.com
Enviada: terça-feira, 12 de maio de 2009 20:55
Para: ccie_voice@onlinestudylist.com
Assunto: [OSL
I Think you have to use Police via MQC. E.g.:
policy-map X
class Y
priority 56
class Z
bandwidth percent 5
police rate 8000
conform-action transmit
exceed-action set-dscp-transmit cs1
violate-action drop
class class-default
Date: Tue, 12 May 2009 10:02:24 -0400
Try this:
send the number to gk like 1999888777. The objetive is match the zone prefix
UCM 1*
voice translation-rule 40
rule 1 /^9001/ /1/
At CM, make your Translation-Pattern strip-off 2#, prefix 9 or 01 to be
9011999888777 (depend on your dial-plan)
Rgsd, Sergio.
Date: Sat, 16
Try to make your integration by Web and there, there is a option to leave Blank
passwd.
Date: Sat, 16 May 2009 09:50:08 +1000
From: jeremy.coo...@gmail.com
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] BLANK PIN/PASSWORD in cue
Hi,
it's very basic question, but I stuck in it!
Hi,
I'm trying to enable on hook transfer at CME.
Does someone know a way to do that?
E.g.
A calls B
B answers
B press transfer and calls C
B talks to C
B hung up and A is connected to C.
Thanks in advance, Sergio.
_
Novo
-consult
This should be the default action.
correct me if my understanding of the requirement is incorrect.
Larry Hadrava
CCIE #12203 CCNP CCNA
Sr. Support Engineer – IPexpert, Inc.
URL: http://www.IPexpert.com
On Thu, May 21, 2009 at 9:05 PM, Sergio Polizer spoli...@hotmail.com wrote:
Hi,
I'm
must put phone B Off Hook because you don’t have
„Transfer” key on „alerting” mode.
Hth,
Cristian
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Sergio Polizer
Sent: 22 May, 2009 4:46 PM
To: cyrus@gmail.com; lhadr
Try to uncheck the box related to automatic configuration on the IPMA user
configurantion page.
Sergio.
Date: Mon, 25 May 2009 13:34:38 -0600
From: jcisc...@gmail.com
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] IPMA issue
hi,
I tried to config IPMA in CCm 4.1.3, when i
I see that you configured VIPM using IP Add but you set the domain-name. How
about if you change the ip domain-name cme.com to the CUE ip add.
Does it fix your issue?
Hth, Sergio.
Date: Tue, 26 May 2009 12:19:55 -0500
From: ccieid...@gmail.com
To: kevho...@cisco.com
CC:
Hi,
Someone can see a way to IPCC be aware of a MWI status of a user? Suppose that
the script may route a call based on the MWI status.
I tried via DB read but I did not find any related variable at CM Sql databese.
Also, I tried via MultTennat MWI and make a Xpattern to Pilot Point, but no
Yes, Robert.
I tried to send the MWI to IPCC and read the ANI/Dnis with a Get call contant
Info. But this does not trigger the Crs application.
Sergio.
Date: Tue, 26 May 2009 22:39:20 +0200
From: rschukne...@gmx.de
Subject: Antw: [OSL | CCIE_Voice] v2 - IPCC aware of MWI
To:
Hi, If you could associate a Label for the secondary lines, a possible
solution could be:
Face Requirement:Phone 1
Line1 : 3001
Line2 : Support 1Phone 2
Line1 : 3002
Line2 : Support 2
ephone-dn 1
number 3001
ephone-dn 2
number 3002
ephone-dn 3
number 3101
ephone-dn 4
number 3102
my thoughts here though! :)
On Thu, May 28, 2009 at 11:23 PM, Sergio Polizer spoli...@hotmail.com wrote:
Hi, If you could associate a Label for the secondary lines, a possible
solution could be:
Face Requirement:
Phone 1
Line1 : 3001
Line2 : Support 1
Phone 2
Line1
.
But, you can not have two labels for the second line in that case.
Cristi
From:
Sergio Polizer [mailto:spoli...@hotmail.com]
Sent: 28 May, 2009 4:55 PM
To: cyrus@gmail.com
Cc: Cristi Radescu; ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice]
GDM configuration
No, MMoH is not considered in location bandwidth.
Could be a wrong MRG of the GW Device Pool?
Date: Thu, 2 Jul 2009 16:20:04 -0400
From: scott.odonn...@gmail.com
To: cristobalpri...@gmail.com
CC: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Questions regarding IOS
I think the shared lines has to be the Barge in feature enable for it.
Sergio.
Date: Mon, 24 Aug 2009 10:36:20 +0200
From: pgciscov...@gmx.net
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Line Remote in Use
Hello!
When is a line Remote in use, so that this text it is
Hi,
I'm trying to block just the calling name via SIP trunk but when I set Calling
Name Presentation to restricted it block the number also. For MGCP and H323
Trunks, no problems.
Something that is missing?
Thanks in advance, Sergio.
[ccie_voice-boun...@onlinestudylist.com] On Behalf Of Sergio Polizer
[spoli...@hotmail.com]
Sent: Sunday, September 27, 2009 9:34 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] SIP Trunk Blocking Calling Name
Hi,
I'm trying to block just the calling name via SIP trunk
Hi,
Suppose the following scenario:
- CAC via RSVP active
- no more bw resources (forced via ip rsvp band 39)
- AAR active
- SNR active for HQPh2
When Br1Ph2 calls HQPh2 just his PSTN number rings. But If I disable SNR via
Mobility soft key, AAR works and HQPh2 rings.
Does someone see it
Check if you have a Line associate to this device.
Date: Tue, 13 Oct 2009 16:13:15 -0700
From: vccie2...@gmail.com
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] No Dialtone on SCCP IP Blue Phone
I see phone registered but no dialtone and it shows Redial button greyed out.
Has
Is it for SCCP phones? How about pagging?
ephone-dn 5
number 3005
pagging
ephone 1
pagging-dn 5
Just make a call to 3005 and the device associated will auto-answer like
Intercom.
Hope that helps, Sergio.
From: mciarfe...@iplogic.com
To: ashfaaq.poonaw...@gmail.com;
In this
scenario, 2 x G729 call over the wan, If we use reservation at 80Kbps we'll
have two calls ringing in the same time
and either blocking three calls. Because, in the worst case, 2 calls already
established
(48Kbps) plus a third call ringing (40kbps) will be 88kbps, so it won't be
Hi, I had a similar problem and I could resolved it adding to my hq-xcoder the
codec g729ar8. Let me know if It work for you.
dspfarm profile 1 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729r8
codec g729br8
maximum sessions 4
associate application SCCP
Regards,
Hello, I'm trying
to access voicemail messages from CUPC without Ldap and DNS. So, I added the
CUPS
hostname at host.txt filename. Desk Phone control is working and It shows me
the Messages and CLID/Time of them, but when I click at the message icon, the
system gives the error Cannot play or
Interesting,
the voicemail audio files are at my PC \Cisco\Unified Personal
Communicator\VoiceMail
CUPC can deleted and mark as read at CUC but cannot play.
I tried a reinstall but no success. Does anybody have any similar problem?
It is related to VOL 1 LAB 13.
From:
Hi Robert,
It works as expected with my 7971 SIP Phones (SIP70.8-4-1S).
I just set Transfer On-Hook Enabled”
to True
From: bobwmcg...@verizon.net
To: ccie_voice@onlinestudylist.com
Date: Mon, 2 Nov 2009 23:03:39 -0500
Subject: [OSL | CCIE_Voice] CUCM Transfers
My HQ has two
Matthew,
You could change MWI numbers that for that specific station. Apply at sip dial
peer pointing to CUE as incomming translation rule.
/MWI_Nsº\(30500\)/ /MWI_Nsº\(30552\)/
From: mjbe...@krollontrack.com
To: ccie_voice@onlinestudylist.com; cisco-voip-boun...@puck.nether.net
Date: Tue,
There is a explanation at SRND CM 7.0 page 657:
When this feature is enabled, Unified CM waits for the remote H.323 device to
send its Terminal
Capabilities Set (TCS) to Unified CM before Unified CM will send its TCS to the
H.323 device. When
the option is disabled, Unified CM does not
I saw that
too . It looks like he made a call from Assistant Primary Line (5002) to
Manager and It was automatic diverted to the Proxy Line (1560).
There is no logic in the real life, but it’s a way to test if we have just two
phones (Manager and Assistant) near you.
From:
I agree. Third bullit should be srr-queue bandwidth shape 10 2 5 5
And If We use auto qos maybe We should map dscp-map queue as well. According
QoS SRND (2-58), DSCP-to-Queue/Threshold maps override CoS-to-Queue/Threshold
maps. Or no mls qos srr-queue output dscp-map.
May Someone give a help
Hi,
Does someone could confirm if is possible to get Call History Presence for
Missed and Received calls when the numbers are globalized?
I'm trying to get it work (Vol 2 Lab4 Q7.3), but It just works for Placed and
Corporate Directory where the number are not globalized ex. 5002 and 1002.
, make sure your presence groups are properly configured to allow
subscriptions from 5002 to 1002 dn
HTH,
On Wed, Jan 27, 2010 at 5:16 PM, Sergio Polizer spoli...@hotmail.com wrote:
Hi,
Does someone could confirm if is possible to get Call History Presence for
Missed and Received calls
Hi Everboby,
Coming Soon Askt to the Expert with
Ben Ng
Get an update on CCIE Voice certification.
Starts February 1, 2010
https://supportforums.cisco.com/community/netpro/ask-the-expert
Hi Harry,
You could use COR.
Sergio.
From: harryshen...@hotmail.com
To: bobwmcg...@verizon.net; ccie_voice@onlinestudylist.com
Date: Fri, 29 Jan 2010 22:18:38 +
Subject: Re: [OSL | CCIE_Voice] Use voice translation-rule to reject outgoing
calls
Hi Bob,
call-block
Hi Ventak,
Check if you have:
voice service voip
allow connections h323 to h323
and allow connections sip to h323, if you have phones Sip.
Also, 10.10.110.3 should be your Loopback IP address.
Sergio.
Date: Sun, 31 Jan 2010 20:11:02 +0530
From: venpa...@yahoo.co.in
To:
Mike,
I'm wondering if you have entered with create profile and proceed with a
reset sip phone after each test?
Thank You,
Sergio.
Date: Tue, 2 Feb 2010 14:00:03 +1100
From: ccievoi...@gmail.com
To: mjbe...@krollontrack.com
CC: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] SIP
Hi,
If you are using calling Party Transformation CSS, try uncheck the Device Pool
Calling Party Transformation CSS at Remote Destination Profile.
If not, try to create a RP for your Remote Destination number and check Use
Calling Party's External Phone Number Mask
HTH, Sergio.
Date: Tue, 9
Hi,
It is based on CTI. So, you need to:
- check if CTIManger service is running on CUCM
- CUCM/End User associate the user to the Standart CTI Enable Group and Specify
Primary Extenssion for end user
- CUPS/Application/CUPC/USer settings, set a CTI Gateway Profile. E.g.
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