Re: [OSL | CCIE_Voice] UCCX/IVR Script Repository

[Brian Chips]

a bit out of topic,what material is best for writing the UCCX scripts?

On Thu, Mar 11, 2010 at 9:06 PM, Tanner Ezell wrote:

> C:\program files\wfavvid\Scripts\Templates
>
> On Thu, Mar 11, 2010 at 1:53 PM, Berry, Matthew J.
>  wrote:
> > Does anyone know of where Cisco’s UCCX/IVR sample script repository is?
> I
> > can’t find it.
> >
> > ___
> > For more information regarding industry leading CCIE Lab training, please
> > visit www.ipexpert.com
> >
> >
>
>
>
> --
> Regards,
> Tanner Ezell
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>



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Re: [OSL | CCIE_Voice] RSVP WITH MLPoFR

[Roger Källberg]

That's the expected behavior. Auto qos won't move the ip rsvp bandwith command. 
That's one of the quirks with auto qos.

Brgds,
Roger Källberg
Unified Communication Consultant
Cygate AB


From: Angel Perez [mailto:gorr...@hotmail.com]
Sent: den 10 mars 2010 19:37
To: osl osl
Subject: [OSL | CCIE_Voice] RSVP WITH MLPoFR

Hello:

I was configurin MLPoFR and LFI on a link between hq and br1, on the serial 
interface I had:

interface Serial0/2/0.202 point-to-point
  ip rsvp bandwidth 64

Calls where progressing as configured (two g729 calls)

Then after apply auto qos voip trust fr-atm new virtual templates and virtual 
access interfaces are created

Then trying to test the policy-map just created and tuned I noticed that I 
could not make calls from hq to br1 (rsvp was rejecting the call)

So I added the following at hq and br1:

interface Virtual-Template200
 ip rsvp bandwidth 64

And the problem get solved

Is this the normal situation? I suppose it is but not 100% sure

Thanks



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Re: [OSL | CCIE_Voice] SIP Hardware Transcoder

[bkvalentine]

Yes it certainly can be.

Did you enable CUBE functionalitu?  The sip phone registers to cucme as a voip 
dialpeer.  The cucm will also talk to cucme as a sip dialpeer.  Make sure you 
allow H323 to SIP, SIP to H323, or SIP to SIP in the gateway as necessary. 

Happy labbing

Brian


Sent via BlackBerry from T-Mobile

-Original Message-
From: Jeff Cotter 
Date: Thu, 11 Mar 2010 16:59:41 
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] SIP Hardware Transcoder

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[OSL | CCIE_Voice] SIP Hardware Transcoder

[Jeff Cotter]

Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM.  
Can't seem to get a call from Call Manager to CME sip phone working.  I can 
call from CME to UCM but not the other way around. Rings but disconnects when 
answered.  Transcoder shows registered in Call manager.  Thanks


Jeff
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Re: [OSL | CCIE_Voice] PSTN Call Distortion Between T1/E1

[Jason Granat]

So I've got this partially figured out. It had to do with the compand-type. E1 
was a-law and T1 was u-law. I set the E1 side for u-law and it sounds correct 
now.

The final thing I am trying to figure out is how to 'trans-compand' (if that is 
the correct term) on the PSTN gateway. As it sits I had to change the 
compand-type between the PSTN and E1 gateway. I don't have experience with 
foreign connectivity so maybe this is the way it is done in the real world but 
I am thinking that perhaps the E1 site may not want or be able to change their 
compand-type, so can it be changed at the PSTN level between a-law and u-law 
locations?

Thanks,

Jason

From: Jason Granat
Sent: Thursday, March 11, 2010 9:46 AM
To: 
Subject: PSTN Call Distortion Between T1/E1

Perhaps this is something simple that I am overlooking but I have the generic 
setup running in my home lab with 3 gateways and one PSTN router. 2 of the 
gateways are T1 and one is E1. The PSTN router is also running CME with a 7960 
to simulate PSTN destinations. Calls from any site to the PSTN phone are fine. 
Calls between T1 sites are fine. Calls between T1 and E1 sites are distorted, 
like the gain is way too high. I tried playing with the gain on the voice-port 
but no luck. I'm not finding much online or in Cisco docs. Any suggestions?

Thanks,

Jason




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Re: [OSL | CCIE_Voice] UC and cme sip integration

[Omotayo]

Hello all,

As anyone been able to get the SIP integration between Unity Connection and
Cme to work? I followed the Proctorlabs Guide

I posted this sometime lat week and revised as advised but keep getting a
reorder tone( Number Unknown) when the message button is pressed
Below is the relevant configuration

voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 no supplementary-service sip moved-temporarily
 no supplementary-service sip refer
 sip
  bind control source-interface Loopback0
  bind media source-interface Loopback0
  registrar server expires max 600 min 60



voice register global
 mode cme
 source-address 10.10.110.3 port 5060
 max-dn 3
 max-pool 6
 authenticate register
 mwi reg-e164
 voicemail 3600
 tftp-path flash:
 create profile sync 0006855418337003
!
voice register dn  1
 number 3002
 call-forward b2bua busy 3600
 call-forward b2bua mailbox 3002
 call-forward b2bua noan 3600 timeout 12
 name br2 phone 2
 no-reg
 label br2 phone 2
 mwi
!
voice register dn  2
 number 3003
 call-forward b2bua busy 3600
 call-forward b2bua mailbox 3003
 call-forward b2bua noan 3600 timeout 12
 name br2 phone 3
 no-reg
 label br2 phone 3
 mwi
!
voice register pool  1
 id mac ..
 type 7941
 number 1 dn 1
 dtmf-relay rtp-nte
 username 3002 password cisco
!
voice register pool  2
 id mac 001F.6C7E.D6FE
 type 7941
 number 1 dn 2
 dtmf-relay rtp-nte
 username 3003 password cisco


dial-peer voice 200 voip
 max-conn 1
 destination-pattern 3600
 session protocol sipv2
 session target ipv4:10.10.210.13
 dtmf-relay rtp-nte
 codec g711ulaw
!
!

telephony-service
  no auto-reg-ephone
 em logout 0:0 0:0 0:0
 max-ephones 8
 max-dn 8
 ip source-address 10.10.202.1 port 2000
 voicemail 3600
 mwi relay
 max-conferences 8 gain -6
 transfer-system full-consult
 transfer-pattern .T
 create cnf-files version-stamp 7960 Mar 10 2010 15:22:39
!
!
ephone-dn  1  dual-line
 number 3001 no-reg primary
 label Br2 pHone 1
 name Br2 Phone 1
 call-forward busy 3600
 call-forward noan 3600 timeout 12
!
!
sip-ua
 mwi-server ipv4:10.10.210.13 expires 3600 port 5060 transport udp
unsolicited
!
!
ephone  1
 device-security-mode none
 mac-address 001E.EC15.996D
 type CIPC
 button  1:1
!

Thanks for the anticipated support
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Re: [OSL | CCIE_Voice] UCCX/IVR Script Repository

[Tanner Ezell]

C:\program files\wfavvid\Scripts\Templates

On Thu, Mar 11, 2010 at 1:53 PM, Berry, Matthew J.
 wrote:
> Does anyone know of where Cisco’s UCCX/IVR sample script repository is?  I
> can’t find it.
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>



-- 
Regards,
Tanner Ezell
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Re: [OSL | CCIE_Voice] Lab 4 AB GK registration problem

[Mike Peterson]

Hi Angel,

Thanks for helping me out with this GK issue. Yes indeed the GW doesn't receive 
the message , that is why we are seeing GRQ and GCF. 
I do have full connectivity b/w HQ/BR2/PUB/SUB .
Below are the ping's you sugested to post.

Thanks a lot in advance for your time and help.

HQ#ping 192.21.66.254->from HQ to loopback of BR2

Type escape sequence to abort.
Sending 5, 100-byte ICMP Echos to 192.21.66.254, timeout is 2 seconds:
!
Success rate is 100 percent (5/5), round-trip min/avg/max = 1/5/12 ms
HQ#


BR2-RTR#ping 192.21.64.254-->from BR2 to loopback of HQ

Type escape sequence to abort.
Sending 5, 100-byte ICMP Echos to 192.21.64.254, timeout is 2 seconds:
!
Success rate is 100 percent (5/5), round-trip min/avg/max = 8/14/32 ms
BR2-RTR#





HQ#ping 192.168.0.11  >from HQ to  CUCM PUB

Type escape sequence to abort.
Sending 5, 100-byte ICMP Echos to 192.168.0.11, timeout is 2 seconds:
!
Success rate is 100 percent (5/5), round-trip min/avg/max = 1/4/8 ms



HQ#ping 192.168.0.12 -> from HQ to CUCM SUB

Type escape sequence to abort.
Sending 5, 100-byte ICMP Echos to 192.168.0.12, timeout is 2 seconds:
!
Success rate is 100 percent (5/5), round-trip min/avg/max = 16/25/40 ms
HQ#


BR2-RTR#ping 192.168.0.11  ->from BR2 CUCM PUB

Type escape sequence to abort.
Sending 5, 100-byte ICMP Echos to 192.168.0.11, timeout is 2 seconds:
!
Success rate is 100 percent (5/5), round-trip min/avg/max = 16/27/44 ms


BR2-RTR#ping 192.168.0.12---> from BR2 to CUCM SUB

Type escape sequence to abort.
Sending 5, 100-byte ICMP Echos to 192.168.0.12, timeout is 2 seconds:
!
Success rate is 100 percent (5/5), round-trip min/avg/max = 20/40/52 ms
BR2-RTR#






From: Angel Perez 
Sent: Thu, March 11, 2010 1:17:17 PM
Subject: RE: [OSL | CCIE_Voice] Lab 4 AB GK registration problem

 Just to verify, can you ping hq loo 0  192.21.64.254 from br2? And br2 loop 
192.21.66.254 from hq?
 
It looks like br2 gw ask for registration GRQ, and then gk try to confirm GCF 
but the gw can't recieve the message

hth 

 Date: Thu, 11 Mar 2010 08:54:30 -0800


Subject: Re: [OSL | CCIE_Voice] Lab 4 AB GK registration problem

 
Hi All,

I did tried your suggestion (to add loopback IP address :
 zone local PL cisco.com 192.21.64.254 ) which does make sense  but it doesn't 
work.
I took a look at "deb ras" and I am seeing only GRQ (a message sent by endpoint 
to GK ) and GCF (A reply from gatekeeper to endpoint
which indicates the transport address of the gatekeeper RAS channel) and I am 
not seeing GRJ (the reject the endpoint request for registration) so something 
I am missing or  I am hitting a BUG!
The "deb gatekeeper main 19" or "deb h225 asn1" still doesn't give me a clue of 
why GK is failing to register.

Once again thanks for your time and help.

Kind Regards,

Mike


Note: This is the change I made:

gatekeeper
 zone local PL cisco.com 192.21.64.254 
 zone prefix PL 1... gw-priority 10 gk-trunk_2
 zone prefix PL 1... gw-priority 9 gk-trunk_1
 zone prefix PL 1... gw-priority 0 BR2-RTR
 zone prefix PL 5... gw-priority 10 gk-trunk_2
 zone prefix PL 5... gw-priority 9 gk-trunk_1
 zone prefix PL 5... gw-priority 0 BR2-RTR
 no shutdown
! 




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[OSL | CCIE_Voice] UCCX/IVR Script Repository

[Berry, Matthew J.]

Does anyone know of where Cisco's UCCX/IVR sample script repository is?  I 
can't find it.
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Re: [OSL | CCIE_Voice] Lab 4 AB GK registration problem

[Angel Perez]

Just to verify, can you ping hq loo 0  192.21.64.254 from br2? And br2 loop 
192.21.66.254 from hq?

 

It looks like br2 gw ask for registration GRQ, and then gk try to confirm GCF 
but the gw can't recieve the message


hth 


Date: Thu, 11 Mar 2010 08:54:30 -0800
From: polobi...@yahoo.com
To: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Lab 4 AB GK registration problem




Hi All,

I did tried your suggestion (to add loopback IP address :
 zone local PL cisco.com 192.21.64.254 ) which does make sense  but it doesn't 
work.
I took a look at "deb ras" and I am seeing only GRQ (a message sent by endpoint 
to GK ) and GCF (A reply from gatekeeper to endpoint
which indicates the transport address of the gatekeeper RAS channel) and I am 
not seeing GRJ (the reject the endpoint request for registration) so something 
I am missing or  I am hitting a BUG!
The "deb gatekeeper main 19" or "deb h225 asn1" still doesn't give me a clue of 
why GK is failing to register.

Once again thanks for your time and help.

Kind Regards,

Mike


Note: This is the change I made:

gatekeeper
 zone local PL cisco.com 192.21.64.254 
 zone prefix PL 1... gw-priority 10 gk-trunk_2
 zone prefix PL 1... gw-priority 9 gk-trunk_1
 zone prefix PL 1... gw-priority 0 BR2-RTR
 zone prefix PL 5... gw-priority 10 gk-trunk_2
 zone prefix PL 5... gw-priority 9 gk-trunk_1
 zone prefix PL 5... gw-priority 0 BR2-RTR
 no shutdown
!




  
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Re: [OSL | CCIE_Voice] PSTN Call Distortion Between T1/E1

[Jason Granat]

I've come across a single post from another individual having the same issue 
from early 2009 but no responses. I've also come across some hints at PCM type, 
but not finding the answer.

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jason Granat
Sent: Thursday, March 11, 2010 9:54 AM
To: Jeff Price (jeffpric); ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] PSTN Call Distortion Between T1/E1

Nope. I have all hardware here in my home lab. No VPN's.

From: Jeff Price (jeffpric) [mailto:jeffp...@cisco.com]
Sent: Thursday, March 11, 2010 9:48 AM
To: Jason Granat; ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] PSTN Call Distortion Between T1/E1

Are you by any chance running a VPN from routers to PSTN?  I've noticed this 
causes distortion some times.

Jeff

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jason Granat
Sent: Thursday, March 11, 2010 9:46 AM
To: 
Subject: [OSL | CCIE_Voice] PSTN Call Distortion Between T1/E1

Perhaps this is something simple that I am overlooking but I have the generic 
setup running in my home lab with 3 gateways and one PSTN router. 2 of the 
gateways are T1 and one is E1. The PSTN router is also running CME with a 7960 
to simulate PSTN destinations. Calls from any site to the PSTN phone are fine. 
Calls between T1 sites are fine. Calls between T1 and E1 sites are distorted, 
like the gain is way too high. I tried playing with the gain on the voice-port 
but no luck. I'm not finding much online or in Cisco docs. Any suggestions?

Thanks,

Jason




http://slash128.com




http://slash128.com




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Re: [OSL | CCIE_Voice] PSTN Call Distortion Between T1/E1

[Jason Granat]

Nope. I have all hardware here in my home lab. No VPN's.

From: Jeff Price (jeffpric) [mailto:jeffp...@cisco.com]
Sent: Thursday, March 11, 2010 9:48 AM
To: Jason Granat; ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] PSTN Call Distortion Between T1/E1

Are you by any chance running a VPN from routers to PSTN?  I've noticed this 
causes distortion some times.

Jeff

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jason Granat
Sent: Thursday, March 11, 2010 9:46 AM
To: 
Subject: [OSL | CCIE_Voice] PSTN Call Distortion Between T1/E1

Perhaps this is something simple that I am overlooking but I have the generic 
setup running in my home lab with 3 gateways and one PSTN router. 2 of the 
gateways are T1 and one is E1. The PSTN router is also running CME with a 7960 
to simulate PSTN destinations. Calls from any site to the PSTN phone are fine. 
Calls between T1 sites are fine. Calls between T1 and E1 sites are distorted, 
like the gain is way too high. I tried playing with the gain on the voice-port 
but no luck. I'm not finding much online or in Cisco docs. Any suggestions?

Thanks,

Jason




http://slash128.com




http://slash128.com
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Re: [OSL | CCIE_Voice] PSTN Call Distortion Between T1/E1

[Jeff Price (jeffpric)]

Are you by any chance running a VPN from routers to PSTN?  I've noticed
this causes distortion some times.

 

Jeff

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jason
Granat
Sent: Thursday, March 11, 2010 9:46 AM
To: 
Subject: [OSL | CCIE_Voice] PSTN Call Distortion Between T1/E1

 

Perhaps this is something simple that I am overlooking but I have the
generic setup running in my home lab with 3 gateways and one PSTN
router. 2 of the gateways are T1 and one is E1. The PSTN router is also
running CME with a 7960 to simulate PSTN destinations. Calls from any
site to the PSTN phone are fine. Calls between T1 sites are fine. Calls
between T1 and E1 sites are distorted, like the gain is way too high. I
tried playing with the gain on the voice-port but no luck. I'm not
finding much online or in Cisco docs. Any suggestions?

 

Thanks,

 

Jason

 





http://slash128.com

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[OSL | CCIE_Voice] PSTN Call Distortion Between T1/E1

[Jason Granat]

Perhaps this is something simple that I am overlooking but I have the generic 
setup running in my home lab with 3 gateways and one PSTN router. 2 of the 
gateways are T1 and one is E1. The PSTN router is also running CME with a 7960 
to simulate PSTN destinations. Calls from any site to the PSTN phone are fine. 
Calls between T1 sites are fine. Calls between T1 and E1 sites are distorted, 
like the gain is way too high. I tried playing with the gain on the voice-port 
but no luck. I'm not finding much online or in Cisco docs. Any suggestions?

Thanks,

Jason




http://slash128.com
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Re: [OSL | CCIE_Voice] Lab 4 AB GK registration problem

[Mike Peterson]

Hi All,

I did tried your suggestion (to add loopback IP address :
 zone local PL cisco.com 192.21.64.254 ) which does make sense  but it doesn't 
work.
I
took a look at "deb ras" and I am seeing only GRQ (a message sent by
endpoint to GK ) and GCF (A reply from gatekeeper to endpoint
which
indicates the transport address of the gatekeeper RAS channel) and I am
not seeing GRJ (the reject the endpoint request for registration) so
something I am missing or  I am hitting a BUG!
The "deb gatekeeper main 19" or "deb h225 asn1" still doesn't give me a clue of 
why GK is failing to register.

Once again thanks for your time and help.

Kind Regards,

Mike


Note: This is the change I made:

gatekeeper
 zone local PL cisco.com 192.21.64.254 
 zone prefix PL 1... gw-priority 10 gk-trunk_2
 zone prefix PL 1... gw-priority 9 gk-trunk_1
 zone prefix PL 1... gw-priority 0 BR2-RTR
 zone prefix PL 5... gw-priority 10 gk-trunk_2
 zone prefix PL 5... gw-priority 9 gk-trunk_1
 zone prefix PL 5... gw-priority 0 BR2-RTR
 no shutdown
!


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Re: [OSL | CCIE_Voice] BARGE

[Angel Perez]

Hi, you are right the help file  is oudated 

 

Anyway it's good to know that the barge feature is limited to g711 streams, to 
mix g729 streams cBarge feature is necessary

 

Thanks Otto

 

 

 


Date: Thu, 11 Mar 2010 08:06:45 -0430
Subject: Re: [OSL | CCIE_Voice] BARGE
From: o...@ipexpert.com
To: gorr...@hotmail.com
CC: ccie_voice@onlinestudylist.com

Hi Angel,

This setting will enable or disable the built-in bridge for all phones that 
support the barge feature (you still have the phone configuration that may 
override this configuration), I think it's outdated in the help,

For a comprehensive list of phones that support the barge feature please take a 
look at the SRND (endpoints chapter):

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/endpnts.html



On Thu, Mar 11, 2010 at 4:48 AM, Angel Perez  wrote:


Hello:
 
It seems to be a limitation on barge feature on phones other than 7940, 7960, 
and 7970 
 
>From ccm help (service parameter):
Built in brige: This parameter determines whether the bridge that is built in 
to Cisco IP Phone models 7940, 7960, and 7970 is enabled
 
So I suppose that for model 7961/7941 and higher the only option is cBarge 
instead of Barge
 
Anyone has seen this before
 
Thanks
 
 



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-- 
Regards,

Otto Sanchez 
CCIE #25592 (Voice) 
Support Engineer - IPexpert, Inc.
URL: http://www.IPexpert.com
  
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Re: [OSL | CCIE_Voice] RSVP WITH MLPoFR

[Angel Perez]

Hi Otto, thanks for your confirmation
 


Date: Thu, 11 Mar 2010 08:48:10 -0430
Subject: Re: [OSL | CCIE_Voice] RSVP WITH MLPoFR
From: o...@ipexpert.com
To: gorr...@hotmail.com
CC: ccie_voice@onlinestudylist.com

Hi Angel,

You are right, the "ip rsvp bandwidth" has to be configured wherever the 
outbound ip interface configuration is, as it will carry the reservation 
messages from one rsvp agent to the other, according to the routing protocol in 
place. In your case this new ip outbound interface is virtual-template200

HTH,

On Wed, Mar 10, 2010 at 2:06 PM, Angel Perez  wrote:



Hello:
 
I was configurin MLPoFR and LFI on a link between hq and br1, on the serial 
interface I had:
 
interface Serial0/2/0.202 point-to-point
  ip rsvp bandwidth 64 
 
Calls where progressing as configured (two g729 calls)
 
Then after apply auto qos voip trust fr-atm new virtual templates and virtual 
access interfaces are created
 
Then trying to test the policy-map just created and tuned I noticed that I 
could not make calls from hq to br1 (rsvp was rejecting the call)
 
So I added the following at hq and br1:
 
interface Virtual-Template200
 ip rsvp bandwidth 64
 
And the problem get solved
 
Is this the normal situation? I suppose it is but not 100% sure
 
Thanks

 
 



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-- 
Regards,

Otto Sanchez 
CCIE #25592 (Voice) 
Support Engineer - IPexpert, Inc.
URL: http://www.IPexpert.com
  
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[OSL | CCIE_Voice] via gatekeeper "invia" key word

[Jeff Cotter]

I am struggling a bit with the invia concept.  I think I understand the 
"outvia".  When I lab this up I find the following.

Invia only applies to calls coming from a remote GK.  In order for call to use 
cube I had to configure the invia key word on the actual remote zone.not on 
the destination zone. Sample config of my invia GK

gk zone local ucm cisco.com 1.1.1.1
gk zone local cube
gk zone local cme
gk zone remote gk2 lab.com 2.2.2.2 invia cube
zone prefixs omitted

So calls coming FROM gk2 destined for either ucm or cme zone used the cube.  If 
I applied the invia key word on either ucm or cme zone directly, the cube was 
not invoked.  This seems to conflict with the proctor guide mock lab 1 
statement "invia command when defining the UCME zone would invoke the cube for 
calls coming in from a remote zone".  In my lab applying invia directly to 
destination zone had no affect and cube was not invoked.

Am I missing something.
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Re: [OSL | CCIE_Voice] RSVP WITH MLPoFR

[Otto Sanchez]

Hi Angel,

You are right, the "ip rsvp bandwidth" has to be configured wherever the
outbound ip interface configuration is, as it will carry the reservation
messages from one rsvp agent to the other, according to the routing protocol
in place. In your case this new ip outbound interface is virtual-template200

HTH,

On Wed, Mar 10, 2010 at 2:06 PM, Angel Perez  wrote:

>  Hello:
>
> I was configurin MLPoFR and LFI on a link between hq and br1, on the serial
> interface I had:
>
> *interface Serial0/2/0.202 point-to-point
>   ip rsvp bandwidth 64 *
>
> Calls where progressing as configured (two g729 calls)
>
> Then after apply *auto qos voip trust fr-atm *new virtual templates and
> virtual access interfaces are created
>
> Then trying to test the policy-map just created and tuned I noticed that I
> could not make calls from hq to br1 (rsvp was rejecting the call)
>
> So I added the following at hq and br1:
>
> *interface Virtual-Template200
>  ip rsvp bandwidth 64*
> **
> And the problem get solved
> **
> Is this the normal situation? I suppose it is but not 100% sure
>
> Thanks
>
>
>
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>


-- 
Regards,

Otto Sanchez
CCIE #25592 (Voice)
Support Engineer - IPexpert, Inc.
URL: http://www.IPexpert.com
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Re: [OSL | CCIE_Voice] BARGE

[Otto Sanchez]

Hi Angel,

This setting will enable or disable the built-in bridge for all phones that
support the barge feature (you still have the phone configuration that may
override this configuration), I think it's outdated in the help,

For a comprehensive list of phones that support the barge feature please
take a look at the SRND (endpoints chapter):

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/endpnts.html


On Thu, Mar 11, 2010 at 4:48 AM, Angel Perez  wrote:

>  Hello:
>
> It seems to be a limitation on barge feature on phones other than 7940,
> 7960, and 7970
>
> From ccm help (service parameter):
> *Built in brige: This parameter determines whether the bridge that is
> built in to Cisco IP Phone models 7940, 7960, and 7970 is enabled*
> **
> So I suppose that for model 7961/7941 and higher the only option is cBarge
> instead of Barge
>
> Anyone has seen this before
>
> Thanks
>
>
>
> --
> ¿Quieres saber qué móvil eres? ¡Descúbrelo aquí!
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>


-- 
Regards,

Otto Sanchez
CCIE #25592 (Voice)
Support Engineer - IPexpert, Inc.
URL: http://www.IPexpert.com
___
For more information regarding industry leading CCIE Lab training, please visit 
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[OSL | CCIE_Voice] BARGE

[Angel Perez]

Hello:

 

It seems to be a limitation on barge feature on phones other than 7940, 7960, 
and 7970 

 

>From ccm help (service parameter):

Built in brige: This parameter determines whether the bridge that is built in 
to Cisco IP Phone models 7940, 7960, and 7970 is enabled

 

So I suppose that for model 7961/7941 and higher the only option is cBarge 
instead of Barge

 

Anyone has seen this before

 

Thanks

 

 
  
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