[OSL | CCIE_Voice] how to dial-by-name in cue

[bruno]

i have configure dial-by-name on cue. but i don't how to dial.
 i configure user br2ph1 for 3001 and br2ph2 for 3002.  AA number is 3100,when 
i dial 3100 ,press 2 ,then i don't know how to dial-by-name?
  
  --
  Best Regards,
Bruno___
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Re: [OSL | CCIE_Voice] WAN QOS Strategy Question.

[CCIEVoiceKP]

This is really old information ... There are no catos switches on the exam ...

Brat --- your approach is sound, however IMO it's only necessary to 'fix' the 
other sub-interface if they specifically call out a bandwidth setting for that 
interface ... If in doubt bring it to the proctor for clarification.

KP


On Dec 2, 2011, at 7:20 PM, Errol Abrahams  wrote:

> 
> Hi All,
> 
> 
> I have stumble onto this website and  it is very good. Check out the quickest 
> way to apply QOS...thnx
> 
> http://pushkarbhatkoti.wordpress.com/category/qos-in-10-minutes/
> 
> Cheers
> 
> Errol
> 
> 
> 
> 
> On Sat, Dec 3, 2011 at 2:24 PM, ccielabrat  wrote:
> To All, 
> 
> I've been trying to figure the best/fastest way to get a WAN QOS requirement 
> completed on exam day.
> 
> I've become very comfortable with Auto-QOS and making the needed tweaks, so 
> Auto-QOS is the way I'm going to use.
> 
> The one piece of the strategy that I'm stilll wondering about is if WAN QOS 
> is specified for only one of the PVC's.
> Auto-QOS will automatically put Frame-relay traffic shaping on the physical 
> interface which has the side effect of leaving the "other" pvc with a 56k PVC 
> speed.
> 
> My solution here is to create a frame-relay map-class with the following 
> parameters.
> 
> map-class frame-relay Not56k
>  frame-relay traffice-rate 1536
> 
> I apply this map-class to the "other" sub-interface/PVC which negates the 56k 
> problem.
> 
> I'm curious if anyone has an opinion on the Pros/Cons of this approach and if 
> it might negate requirements somehow.
> 
> 
> 
> 
> ___
> 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 
> www.PlatinumPlacement.com
> 
> ___
> 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 
> www.PlatinumPlacement.com
___
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Re: [OSL | CCIE_Voice] Introducing myself

[Anthony Alba]

Bonjour Nicholas
I am in the same position, RS trying to move to Voice, I have just passed
the written.
I have also built a lab and I use rack rentals to see the configurations
and deliberate 'bugs'.

My lessons learnt so far for home rack:
** use Intel for your VMware server; the versions of the servers don't
install nice on AMD
** use Unity Connection 8 for home VPIM, as VPIM is in the demo license. I
am running both CUC 7 and 8. I had problems installing 8 on VMware; in the
end I gave it unlimited memory during install and reduced to 3gb after
install.
** try to get at least a pair of 7965 so you see the actual lab phone and
needed to test iLBC tasks
** if you access your rack with local hardware phones to your pod by l2l
VPN you can get multicast moh to work over a gre tunnel to your home
phones. Since you've done RS you'll know what I mean: switch to PIM sparse
mode. I don't use another router at home as the tunnel peer but just a
Linux box and Xorp. I use HQ router as tunnel endpoint and PIM RP. It works
very well.
** if your phones are separated by VPN from servers always set up a local
TFTP servers for firmware and use loadServer in CUCM to tell phones to grab
local firmware. For CME phones I switch firmware using CUCM and loadServer
first. TFTP over the WAN is very very slow.



Anthony




On Friday, December 2, 2011, Nicolas MICHEL  wrote:
> Hey Michael.
> Thanks again for all the help provided with the CCIE RS when I was
studying for it :)
> How far from the CCIE Voice are you now ?
> I m just starting, building a phones lab and then I'll be using IPX and
some other vendors as well I guess 
> I remember you were building a rack with friends, how far are you from
there ? :)
>
> Seeing that this Mailing list is far more active than the RS one ! Cheers
!!
>
> Nic
>
>
>
> 2011/12/2 Michael Miller 
>>
>> Hello Nicholas,
>>
>> Its nice to see some familiar faces from the RS OSL boards. =)
>>
>> Congrats on passing the RS, and good luck on the Voice!
>>
>> Thanks,
>>
>> Michael
>>
>> On Fri, Dec 2, 2011 at 11:30 AM, Emanuel Damasceno 
wrote:
>>>
>>> Welcome Michel!
>>>
>>> You will see this is just a nice journey as RS was for you. I am the
opposite, I started into Voice, and when I get my CCIE, I will start on my
CCNP, and CCIE RS... =)
>>>
>>> Welcome to the UC world. I really love it and I am hoping you will love
it too!
>>> Best regards, brother.
>>> Emanuel Damasceno
>>>
>>>
>>>
>>>
>>> On Fri, Dec 2, 2011 at 8:11 AM, Nicolas MICHEL 
wrote:

 Hey There guys.
 I'm a french network engineer mainly focused into RS but as of now I m
starting to deploy UC solutions and so far so good I like it.
 This is why I decided to pursue my 2nd CCIE into Voice and can't wait
to be there yet :)
 I actually finished the CCNA book and the CBT nuggets for that series
and now digging into CCNP stuff.
 I got CVOICE, CIPT1,CIPT2, Presence, CUCM+unity, Unity books. and then
I ll start to read the SRND which looks awesome.
 I'm also building a lab to use some remote racks.
 If you guys have any advices, I d be glad to hear them :P
 Thanks for your help and cant wait to have the knowledge to ask
question and answer on the OSL :)
 Nic

 --
 Nicolas MICHEL
 Ingenieur Réseaux CCIE #29410






 ___
 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
www.PlatinumPlacement.com
>>>
>>>
>>> ___
>>> 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
www.PlatinumPlacement.com
>>
>
>
>
> --
> Nicolas MICHEL
> Ingenieur Réseaux CCIE #29410
> Tel: +33 6 08 72 75 97
>
>
>
>
>
___
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Re: [OSL | CCIE_Voice] WAN QOS Strategy Question.

[Errol Abrahams]

Hi All,


I have stumble onto this website and  it is very good. Check out the
quickest way to apply QOS...thnx

*http://pushkarbhatkoti.wordpress.com/category/qos-in-10-minutes/*

Cheers

Errol




On Sat, Dec 3, 2011 at 2:24 PM, ccielabrat  wrote:

> To All,
>
> I've been trying to figure the best/fastest way to get a WAN QOS
> requirement completed on exam day.
>
> I've become very comfortable with Auto-QOS and making the needed tweaks,
> so Auto-QOS is the way I'm going to use.
>
> The one piece of the strategy that I'm stilll wondering about is if WAN
> QOS is specified for only one of the PVC's.
> Auto-QOS will automatically put Frame-relay traffic shaping on the
> physical interface which has the side effect of leaving the "other" pvc
> with a 56k PVC speed.
>
> My solution here is to create a frame-relay map-class with the following
> parameters.
>
> map-class frame-relay Not56k
>  frame-relay traffice-rate 1536
>
> I apply this map-class to the "other" sub-interface/PVC which negates the
> 56k problem.
>
> I'm curious if anyone has an opinion on the Pros/Cons of this approach and
> if it might negate requirements somehow.
>
>
>
>
> ___
> 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
> www.PlatinumPlacement.com
>
___
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www.ipexpert.com

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] DSPFarms for SRST

[Ken Wyan]

Ya , definitely I"ll try this when possible.

But unluckily I am unable to test this with remote racks , which I am using
currently for preparation.

Let's wait for an input from an expert in SRST.

Thanks




On Sat, Dec 3, 2011 at 8:59 AM, Bill Lake  wrote:

> You might be right but Cisco is usually pretty good when they do these
> videos.  Why not try it and see.
>
> On Fri, Dec 2, 2011 at 9:26 PM, Ken Wyan  wrote:
>
>> Hi,
>>
>> If we configure a dhcp pool on gateway , (some) phones will not get IP
>> addresses from CUCM dhcp pool. Then there will be ip address conflicts
>> unless we divide ip range between CUCM & gateway.
>>
>> After resolving this , we have to provide a prioritized list of tftp
>> servers under option 150 in gateway router ( CUCM sub , CUCM pub , gateway
>> ).
>>
>> As mentioned in video if we give only gateway ip for option 150 , some
>> phones will never able to download conf file from CUCM.
>>
>> I think dhcp / tftp redundancy part in video is wrong.
>>
>> On Sat, Dec 3, 2011 at 8:43 AM, Bill Lake  wrote:
>>
>>> Thanks guys, I thought it was a good video and does a good job
>>>
>>> As for DHCP, I think this is the age old question, Centralized vs
>>> Distributed. Each has their benefits but I believe that your phones might
>>> have some trouble if the WAN link is down and they need an IP and your DHCP
>>> server is centrally located. Learn both ways as I am sure you will be
>>> tested on it.
>>>
>>>
>>> On Fri, Dec 2, 2011 at 8:40 PM, Ken Wyan  wrote:
>>>
 Nice Link & watched the video fully.

 As mentioned in this QLM , we should configure SRST router as dhcp
 server & provide gateway address as tftp (option 150) as well. Reason
 quoted is during wan failure phones will not have access to CUCM which
 provides DHCP & TFTP services.

 I have never configured dhcp in Router  ( & used helper address to
 remote dhcp server ) & srst works fine & I have a doubt if we should
 provide redundant DHCP & TFTP & is it really possible.






 On Sat, Dec 3, 2011 at 4:52 AM, Bill Lake  wrote:

> You might want to watch this
>
>
>
> https://www.cisco.com/web/learning/le31/le46/cln/qlm/CCVP/cipt1/cucme_as_srst/player.html
>
>
>
>
> On Fri, Dec 2, 2011 at 2:09 PM, Gurpreet Singh Kukreja <
> tycoononway1...@gmail.com> wrote:
>
>> Hi,
>>
>> You can configure the dspfarms under telephony-service as far as i
>> know but not under call-m-f. Under telephony service, you can use the
>> sdsfarm commands to do so.
>>
>> - Gurpreet
>>
>>
>> On Fri, Dec 2, 2011 at 11:22 AM, datucha123 datucha123 <
>> datucha...@gmail.com> wrote:
>>
>>> Hello,
>>>
>>> I can configure the Transcoding for CUCME SRST.
>>>
>>> But as I guess the call-manager-fallback does not support
>>> registering the dspfarm profiles. So no Transcoder or any other profile 
>>> is
>>> supported in Traditional SRST.
>>>
>>> Am I right?
>>>
>>> ___
>>> 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
>>> www.PlatinumPlacement.com 
>>>
>>
>>
>> ___
>> 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
>> www.PlatinumPlacement.com 
>>
>
>
> ___
> 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
> www.PlatinumPlacement.com 
>


>>>
>>
>
___
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 
www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] DSPFarms for SRST

[Bill Lake]

You might be right but Cisco is usually pretty good when they do these
videos.  Why not try it and see.

On Fri, Dec 2, 2011 at 9:26 PM, Ken Wyan  wrote:

> Hi,
>
> If we configure a dhcp pool on gateway , (some) phones will not get IP
> addresses from CUCM dhcp pool. Then there will be ip address conflicts
> unless we divide ip range between CUCM & gateway.
>
> After resolving this , we have to provide a prioritized list of tftp
> servers under option 150 in gateway router ( CUCM sub , CUCM pub , gateway
> ).
>
> As mentioned in video if we give only gateway ip for option 150 , some
> phones will never able to download conf file from CUCM.
>
> I think dhcp / tftp redundancy part in video is wrong.
>
> On Sat, Dec 3, 2011 at 8:43 AM, Bill Lake  wrote:
>
>> Thanks guys, I thought it was a good video and does a good job
>>
>> As for DHCP, I think this is the age old question, Centralized vs
>> Distributed. Each has their benefits but I believe that your phones might
>> have some trouble if the WAN link is down and they need an IP and your DHCP
>> server is centrally located. Learn both ways as I am sure you will be
>> tested on it.
>>
>>
>> On Fri, Dec 2, 2011 at 8:40 PM, Ken Wyan  wrote:
>>
>>> Nice Link & watched the video fully.
>>>
>>> As mentioned in this QLM , we should configure SRST router as dhcp
>>> server & provide gateway address as tftp (option 150) as well. Reason
>>> quoted is during wan failure phones will not have access to CUCM which
>>> provides DHCP & TFTP services.
>>>
>>> I have never configured dhcp in Router  ( & used helper address to
>>> remote dhcp server ) & srst works fine & I have a doubt if we should
>>> provide redundant DHCP & TFTP & is it really possible.
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Sat, Dec 3, 2011 at 4:52 AM, Bill Lake  wrote:
>>>
 You might want to watch this



 https://www.cisco.com/web/learning/le31/le46/cln/qlm/CCVP/cipt1/cucme_as_srst/player.html




 On Fri, Dec 2, 2011 at 2:09 PM, Gurpreet Singh Kukreja <
 tycoononway1...@gmail.com> wrote:

> Hi,
>
> You can configure the dspfarms under telephony-service as far as i
> know but not under call-m-f. Under telephony service, you can use the
> sdsfarm commands to do so.
>
> - Gurpreet
>
>
> On Fri, Dec 2, 2011 at 11:22 AM, datucha123 datucha123 <
> datucha...@gmail.com> wrote:
>
>> Hello,
>>
>> I can configure the Transcoding for CUCME SRST.
>>
>> But as I guess the call-manager-fallback does not support registering
>> the dspfarm profiles. So no Transcoder or any other profile is supported 
>> in
>> Traditional SRST.
>>
>> Am I right?
>>
>> ___
>> 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
>> www.PlatinumPlacement.com 
>>
>
>
> ___
> 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
> www.PlatinumPlacement.com 
>


 ___
 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
 www.PlatinumPlacement.com 

>>>
>>>
>>
>
___
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 
www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] Need some help on a Cisco 2811 Router...........................Please help!!

[Bill Lake]

what version of IOS do you have on this router?

On Fri, Dec 2, 2011 at 5:20 PM, Errol Abrahams wrote:

> Hi All,
>
> I have a Cisco 2811 router with a VWIC-2MFT-T1 card and during bootup
> time, only the one channel T1 0/1/0 becomes active. The other channel which
> is T1 0/1/1 doesn't respond, but yet the 'AL' led on the card stays on. If
> I shut down the controller T1 0/1/0 then this channel responds but not the
> other one. Is there a command on the Cisco 2811 to active the other channel.
>
> I would appreciate your responds regarding this matter..thnx.
>
> Cheers
>
> Errol
>
> ___
> 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
> www.PlatinumPlacement.com
>
___
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 
www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] DSPFarms for SRST

[Ken Wyan]

Hi,

If we configure a dhcp pool on gateway , (some) phones will not get IP
addresses from CUCM dhcp pool. Then there will be ip address conflicts
unless we divide ip range between CUCM & gateway.

After resolving this , we have to provide a prioritized list of tftp
servers under option 150 in gateway router ( CUCM sub , CUCM pub , gateway
).

As mentioned in video if we give only gateway ip for option 150 , some
phones will never able to download conf file from CUCM.

I think dhcp / tftp redundancy part in video is wrong.

On Sat, Dec 3, 2011 at 8:43 AM, Bill Lake  wrote:

> Thanks guys, I thought it was a good video and does a good job
>
> As for DHCP, I think this is the age old question, Centralized vs
> Distributed. Each has their benefits but I believe that your phones might
> have some trouble if the WAN link is down and they need an IP and your DHCP
> server is centrally located. Learn both ways as I am sure you will be
> tested on it.
>
>
> On Fri, Dec 2, 2011 at 8:40 PM, Ken Wyan  wrote:
>
>> Nice Link & watched the video fully.
>>
>> As mentioned in this QLM , we should configure SRST router as dhcp server
>> & provide gateway address as tftp (option 150) as well. Reason quoted is
>> during wan failure phones will not have access to CUCM which provides DHCP
>> & TFTP services.
>>
>> I have never configured dhcp in Router  ( & used helper address to remote
>> dhcp server ) & srst works fine & I have a doubt if we should provide
>> redundant DHCP & TFTP & is it really possible.
>>
>>
>>
>>
>>
>>
>> On Sat, Dec 3, 2011 at 4:52 AM, Bill Lake  wrote:
>>
>>> You might want to watch this
>>>
>>>
>>>
>>> https://www.cisco.com/web/learning/le31/le46/cln/qlm/CCVP/cipt1/cucme_as_srst/player.html
>>>
>>>
>>>
>>>
>>> On Fri, Dec 2, 2011 at 2:09 PM, Gurpreet Singh Kukreja <
>>> tycoononway1...@gmail.com> wrote:
>>>
 Hi,

 You can configure the dspfarms under telephony-service as far as i know
 but not under call-m-f. Under telephony service, you can use the sdsfarm
 commands to do so.

 - Gurpreet


 On Fri, Dec 2, 2011 at 11:22 AM, datucha123 datucha123 <
 datucha...@gmail.com> wrote:

> Hello,
>
> I can configure the Transcoding for CUCME SRST.
>
> But as I guess the call-manager-fallback does not support registering
> the dspfarm profiles. So no Transcoder or any other profile is supported 
> in
> Traditional SRST.
>
> Am I right?
>
> ___
> 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
> www.PlatinumPlacement.com 
>


 ___
 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
 www.PlatinumPlacement.com 

>>>
>>>
>>> ___
>>> 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
>>> www.PlatinumPlacement.com 
>>>
>>
>>
>
___
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 
www.PlatinumPlacement.com

[OSL | CCIE_Voice] WAN QOS Strategy Question.

[ccielabrat]

To All,

I've been trying to figure the best/fastest way to get a WAN QOS
requirement completed on exam day.

I've become very comfortable with Auto-QOS and making the needed tweaks, so
Auto-QOS is the way I'm going to use.

The one piece of the strategy that I'm stilll wondering about is if WAN QOS
is specified for only one of the PVC's.
Auto-QOS will automatically put Frame-relay traffic shaping on the physical
interface which has the side effect of leaving the "other" pvc with a 56k
PVC speed.

My solution here is to create a frame-relay map-class with the following
parameters.

map-class frame-relay Not56k
 frame-relay traffice-rate 1536

I apply this map-class to the "other" sub-interface/PVC which negates the
56k problem.

I'm curious if anyone has an opinion on the Pros/Cons of this approach and
if it might negate requirements somehow.
___
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 
www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] DSPFarms for SRST

[Bill Lake]

Thanks guys, I thought it was a good video and does a good job

As for DHCP, I think this is the age old question, Centralized vs
Distributed. Each has their benefits but I believe that your phones might
have some trouble if the WAN link is down and they need an IP and your DHCP
server is centrally located. Learn both ways as I am sure you will be
tested on it.


On Fri, Dec 2, 2011 at 8:40 PM, Ken Wyan  wrote:

> Nice Link & watched the video fully.
>
> As mentioned in this QLM , we should configure SRST router as dhcp server
> & provide gateway address as tftp (option 150) as well. Reason quoted is
> during wan failure phones will not have access to CUCM which provides DHCP
> & TFTP services.
>
> I have never configured dhcp in Router  ( & used helper address to remote
> dhcp server ) & srst works fine & I have a doubt if we should provide
> redundant DHCP & TFTP & is it really possible.
>
>
>
>
>
>
> On Sat, Dec 3, 2011 at 4:52 AM, Bill Lake  wrote:
>
>> You might want to watch this
>>
>>
>>
>> https://www.cisco.com/web/learning/le31/le46/cln/qlm/CCVP/cipt1/cucme_as_srst/player.html
>>
>>
>>
>>
>> On Fri, Dec 2, 2011 at 2:09 PM, Gurpreet Singh Kukreja <
>> tycoononway1...@gmail.com> wrote:
>>
>>> Hi,
>>>
>>> You can configure the dspfarms under telephony-service as far as i know
>>> but not under call-m-f. Under telephony service, you can use the sdsfarm
>>> commands to do so.
>>>
>>> - Gurpreet
>>>
>>>
>>> On Fri, Dec 2, 2011 at 11:22 AM, datucha123 datucha123 <
>>> datucha...@gmail.com> wrote:
>>>
 Hello,

 I can configure the Transcoding for CUCME SRST.

 But as I guess the call-manager-fallback does not support registering
 the dspfarm profiles. So no Transcoder or any other profile is supported in
 Traditional SRST.

 Am I right?

 ___
 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
 www.PlatinumPlacement.com 

>>>
>>>
>>> ___
>>> 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
>>> www.PlatinumPlacement.com 
>>>
>>
>>
>> ___
>> 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
>> www.PlatinumPlacement.com 
>>
>
>
___
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 
www.PlatinumPlacement.com

[OSL | CCIE_Voice] SNUR requires 'Urgent Priority' on globalized translation pattern \+.! ????

[Anthony Alba]

Hello,

Mobile Connect (SNUR) issue:
** using E.164 for remote destination e.g. +12123941234
** using globalized dial plan with one route pattern \+.!
** using one translation pattern \+.! (for plus dialing from directory)
whose CSS sees the global route pattern.

I do not want the devices to see the route pattern directly; every dialed
number goes via a translation pattern.

If the \+.! translation pattern does not have 'Urgent Priority' then mobile
connect cannot route out using this pattern.
Is this expected behaviour?

Actually, I do not want the translation pattern \+.! to have 'Urgent
Priority' because this causes issues with SIP phones and + dialing from
directory.

Is there some reference to this behaviour of SNUR?

Tks.

Anthony
___
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www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] DSPFarms for SRST

[Ken Wyan]

Nice Link & watched the video fully.

As mentioned in this QLM , we should configure SRST router as dhcp server &
provide gateway address as tftp (option 150) as well. Reason quoted is
during wan failure phones will not have access to CUCM which provides DHCP
& TFTP services.

I have never configured dhcp in Router  ( & used helper address to remote
dhcp server ) & srst works fine & I have a doubt if we should provide
redundant DHCP & TFTP & is it really possible.






On Sat, Dec 3, 2011 at 4:52 AM, Bill Lake  wrote:

> You might want to watch this
>
>
>
> https://www.cisco.com/web/learning/le31/le46/cln/qlm/CCVP/cipt1/cucme_as_srst/player.html
>
>
>
>
> On Fri, Dec 2, 2011 at 2:09 PM, Gurpreet Singh Kukreja <
> tycoononway1...@gmail.com> wrote:
>
>> Hi,
>>
>> You can configure the dspfarms under telephony-service as far as i know
>> but not under call-m-f. Under telephony service, you can use the sdsfarm
>> commands to do so.
>>
>> - Gurpreet
>>
>>
>> On Fri, Dec 2, 2011 at 11:22 AM, datucha123 datucha123 <
>> datucha...@gmail.com> wrote:
>>
>>> Hello,
>>>
>>> I can configure the Transcoding for CUCME SRST.
>>>
>>> But as I guess the call-manager-fallback does not support registering
>>> the dspfarm profiles. So no Transcoder or any other profile is supported in
>>> Traditional SRST.
>>>
>>> Am I right?
>>>
>>> ___
>>> 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
>>> www.PlatinumPlacement.com 
>>>
>>
>>
>> ___
>> 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
>> www.PlatinumPlacement.com 
>>
>
>
> ___
> 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
> www.PlatinumPlacement.com 
>
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Re: [OSL | CCIE_Voice] Need Help with MoH

[Mohd Baqari]

Good news ... I was using IOS 12.4(24)T1 which was matching bug IOS

Regards,
Mohammed Al Baqari

Sent from my iPhone

On Dec 2, 2011, at 8:06 PM, datucha123 datucha123  wrote:

> I have tested it in my Lab, the the MoH was working without MTP either 
> through Xcoder or G729 enabled in IPVMS service.
>  
> So that BUG does not effect my CUCM
> 
> On Thu, Dec 1, 2011 at 10:13 PM, datucha123 datucha123  
> wrote:
> 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 the MoH was transcoded :)
> 
> But the BUG you have sent, is very intestring, so I will also test it 
> tomorrow.
> 
> 
> On Thu, Dec 1, 2011 at 7:57 PM, Mohammed Al Baqari  
> wrote:
> 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  
> 
> G0 L 85   N   ORG T4 g729r8  VOIPP 
> 142.2.64.254:17672  
> 
> G0 L 86   N   ORG T4 g729r8  VOIPP 
> 142.2.64.254:17502
> 
> G0 L 88   N   ORG T4 g711ulawVOIPP  
> 0.0.0.0:0
> 
>  
> 
> The interesting part is this. It wasn’t working without MTP (either I am 
> using g711 on MOH which needs XCODE or even if I change MOH to G729 which 
> doesn’t require XCODE). Finally I found that I am hitting this bug. So it’s a 
> must to use MTP over SIP trunk for MOH to work either with XCODE or not.
> 
>  
> 
> CSCso85618 Bug Details
> 
>  
> 
> Top of Form
> 
> No Audio when Call is put on hold by remote party over Sip Trunk
> 
> Symptoms: 
> 
> 1. The MOH does not work between CM and CME. 
> 
> 2. There is no audio on the CME endpoint when the remote CCM party resumes the
> call on hold, conferences, or transfers with another CCM endpoint (scenario: 
> CME - CUBE - CCM).
> 
> Conditions: Symptom 1 is observed if the phone registered to the CME is put on
> hold by the CM, then the CME phone does not hear the MOH.
> 
> Symptom 2 is observed if the CCM endpoint does a conference, hold, or 
> transfer.
> 
> Workaround: Use an MTP.
> 
> Regards,
> 
> Mohammed Al Baqari
> 
>  
> 
> From: ccie_voice-boun...@onlinestudylist.com 
> [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of datucha123 
> datucha123
> Sent: Wednesday, November 30, 2011 10:59 PM
> To: ccie_voice@onlinestudylist.com
> Subject: [OSL | CCIE_Voice] Need Help with MoH
> 
>  
> 
> Hello,
> 
>  
> 
> I am facing a Problem with MoH over the SIP/H323 Trunk with MTP.
> 
>  
> 
> I know, that the multicast MoH is not supported when the MTP is check in the 
> H323 gateway, but I have a problem with Unicast MoH with transcoding.
> 
>  
> 
> But I have the following configuration.
> 
>  
> 
> Only G711U is activated for MoH (default).
> 
>  
> 
> I have a SIP Trunk towards CUCME, this SIP Trunk is in G729 Region, so that 
> this Region is using G729 to CUCM IP Phones, and to MoH regions as well:
> 
>  
> 
> So I have
> 
> HQ_Region  -  G711 Inside and G729 to SIP_Trunk_Region.
> 
> SIP_Trunk_Region  - G711 inside and G729 to HQ_Region.
> 
>  
> 
> So the SIP Trunk is assigned the SIP_Trunk_Region (Through DP) and the IP 
> Phones along with the MoH Server are assigned the HQ_Region.
> 
>  
> 
> Also I have an G729 MTP on a Router, which is assigned the MTP_Region (Which 
> is using G729 to all other Regions).
> 
> Transcoder is assigne the HQ_Region.
> 
>  
> 
> So when I call the SIP Trunk (CUCME IP Phones) the call is successful and 
> negotiated codec is G729, and MTP is also allocated, as I have checked the 
> MTP Required Checkbox.
> 
>  
> 
> But when I press the Hold key on the CUCM IP Phone, the CUCME IP Phone hears 
> a Beeps instead of MoH Music.
> 
>  
> 
> I cannot get how to make transcoder to transcode the MoH from G711 to G729, 
> so that the MTP will be also used on the SIP Trunk.
> 
>  
> 
> BTW, when I remove the MTP Required from the SIP Trunk, the MoH is 
> transcoded, and the remote IP Phone (CUCME) can hear the music.
> 
>  
> 
> But I want to have an MTP checked as well, and in this case the MoH music 
> does not play. (Only Beeps).
> 
>  
> 
> Please help me.
> 
>  
> 
>  
> 
>  
> 
> 
> 
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Re: [OSL | CCIE_Voice] CCNA Voice certification preparation

[Matt Just]

Tom,

 

I have fixed the CCNA & CCNP OSL forums today so feel free to post in the
forums as our instructors and myself will be answering questions and
providing support.

 

Enjoy!

 

Matt Just - IPexpert, Inc.

CTO - Chief Technical Officer

Email:   mj...@ipexpert.com

Office: (810) 326-1444 Ext: 331

Mobile +1 (810) 406-7218

www.ipexpert.com/catalog

  Description: Description: facebook
   

 

This e-mail and any files transmitted with it are IPexpert property, are
confidential, and are intended solely for the use of the individual or
entity to whom this email is addressed. If you are not one of the named
recipient (s) or otherwise have reason to believe that you have received
this message in error, please notify the sender and delete this message
immediately from your computer. Any other use, retention, dissemination,
forwarding, printing, or copying of this e-mail is strictly prohibited.

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Tom Ribbens
Sent: Friday, December 02, 2011 7:22 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] CCNA Voice certification preparation

 

Hi,

 

I realize this is slightly off topic here, but the list for CCNA Voice
bounces. Haven't tried the CCNP Voice list, but as the archives are empty, I
assume the same problem.

 

I took the 640-461 ICOMM exam towards CCNA Voice a couple of weeks ago, and
unfortunately failed. I had the impression a lot of topics were not covered
in the official cert guide from Cisco Press. Are there any resources to
prepare for this exam which are better suited?

 

Sorry again for being slightly off topic.

 

Regards,

 

Tom

<><><><>___
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Re: [OSL | CCIE_Voice] Strange behavior on MGCP gateways

[Matthew Saskin]

I'd look a bit further, the "silence" is not causing the call to terminate
- what about calls being muted, etc.  What do the q931 debugs show when the
call gets disconnected?


On Thu, Dec 1, 2011 at 11:02 PM, ccielabrat  wrote:

> 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 on any of my phones , so
> when I answer with the answer softkey , the phone is off hook and sending
> "dead air" packets.
> Somehow , mgcp see this as an error condition after two minutes or so and
> kills the call.
>
> If the call is on speaker, I guess it picks up enough noise to keep the
> call from being considered inactive.
>
> Weird.
>
>
> ___
> 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
> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] DSPFarms for SRST

[Gurpreet Singh Kukreja]

Nice Link for a quick overview Bill.


On Fri, Dec 2, 2011 at 6:22 PM, Bill Lake  wrote:

> You might want to watch this
>
>
>
> https://www.cisco.com/web/learning/le31/le46/cln/qlm/CCVP/cipt1/cucme_as_srst/player.html
>
>
>
>
> On Fri, Dec 2, 2011 at 2:09 PM, Gurpreet Singh Kukreja <
> tycoononway1...@gmail.com> wrote:
>
>> Hi,
>>
>> You can configure the dspfarms under telephony-service as far as i know
>> but not under call-m-f. Under telephony service, you can use the sdsfarm
>> commands to do so.
>>
>> - Gurpreet
>>
>>
>> On Fri, Dec 2, 2011 at 11:22 AM, datucha123 datucha123 <
>> datucha...@gmail.com> wrote:
>>
>>> Hello,
>>>
>>> I can configure the Transcoding for CUCME SRST.
>>>
>>> But as I guess the call-manager-fallback does not support registering
>>> the dspfarm profiles. So no Transcoder or any other profile is supported in
>>> Traditional SRST.
>>>
>>> Am I right?
>>>
>>> ___
>>> 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
>>> www.PlatinumPlacement.com
>>>
>>
>>
>> ___
>> 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
>> www.PlatinumPlacement.com
>>
>
>
___
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Re: [OSL | CCIE_Voice] DSPFarms for SRST

[Bill Lake]

You might want to watch this


https://www.cisco.com/web/learning/le31/le46/cln/qlm/CCVP/cipt1/cucme_as_srst/player.html




On Fri, Dec 2, 2011 at 2:09 PM, Gurpreet Singh Kukreja <
tycoononway1...@gmail.com> wrote:

> Hi,
>
> You can configure the dspfarms under telephony-service as far as i know
> but not under call-m-f. Under telephony service, you can use the sdsfarm
> commands to do so.
>
> - Gurpreet
>
>
> On Fri, Dec 2, 2011 at 11:22 AM, datucha123 datucha123 <
> datucha...@gmail.com> wrote:
>
>> Hello,
>>
>> I can configure the Transcoding for CUCME SRST.
>>
>> But as I guess the call-manager-fallback does not support registering the
>> dspfarm profiles. So no Transcoder or any other profile is supported in
>> Traditional SRST.
>>
>> Am I right?
>>
>> ___
>> 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
>> www.PlatinumPlacement.com
>>
>
>
> ___
> 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
> www.PlatinumPlacement.com
>
___
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[OSL | CCIE_Voice] Need some help on a Cisco 2811 Router...........................Please help!!

[Errol Abrahams]

Hi All,

I have a Cisco 2811 router with a VWIC-2MFT-T1 card and during bootup time,
only the one channel T1 0/1/0 becomes active. The other channel which is T1
0/1/1 doesn't respond, but yet the 'AL' led on the card stays on. If I shut
down the controller T1 0/1/0 then this channel responds but not the other
one. Is there a command on the Cisco 2811 to active the other channel.

I would appreciate your responds regarding this matter..thnx.

Cheers

Errol
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Re: [OSL | CCIE_Voice] Need some help with GK/CUBE Remote Zone call debugging.

[Chris Martin]

In your situation you have wait for TCS unchecked I believe.  So that means
that CUCM will send the TCS, and you should see the message in debug H245
asn1.  Now your call is probally failing with error 127, interworking
error.  You can confirm this through "debug voice ccapi inout" looking for
disconnect cause code.  Also a "show h323 gateway" will show you a nice set
of statistics and failure codes.  The reason for this is you have a G711
dial-peer trying to communicate with a g729r8 dial-peer internally in that
gateway.

So how to make UCM -> G711 -> CUBE -> G729 -> CME work is setup a local
transcoder, create your profile and register it locally to CME.  Then CUBE
will invoke the transcoder between the two dial-peers and you will see all
the other TCS messages, as the internal error is corrected.  (just make
sure the transcoder is registered locally to CME and NOT CUCM)

I personally found h323 debugs one of the hardest sections to get
comfortable reading through and understanding, don't get discouraged.

HTH,
Chris

On Fri, Dec 2, 2011 at 2:59 PM, ccielabrat  wrote:

> Hi Chris,
> So is CUBE waiting for CUCM to send TCS ?
> And holding up the whole process?
>
> Also, I guess this means the the BBGK is waiting for TCS from CUBE.
> Is CUBE sending TCS independent of the CUCM call leg? or does it wait for
> CUCM TCS before sending TCS to BBGK?
>
>
>
>
> On Fri, Dec 2, 2011 at 3:08 PM, Chris Martin  wrote:
>
>> Because the IOS gateway trunks follow the rfc that the initiator will
>> send TCS, while by default CUCM waits for the TCS from the end point.  Per
>> the SRND this was done for potential video issues, so either you may have
>> video problems with certain endpoints or voice issues with gateways
>> trunks.  You can review this on CUCM SRND, search for TCS.
>>
>> HTH,
>> Chris
>>
>>
>> On Fri, Dec 2, 2011 at 12:45 PM, datucha123 datucha123 <
>> datucha...@gmail.com> wrote:
>>
>>> By the way, why the BBGK endpoint does not send the TCS?
>>>
>>>
>>>
>>>
>>> On Fri, Dec 2, 2011 at 10:02 PM, ccielabrat wrote:
>>>
 Hi Ash,

 Thanks very much for taking the time to reply.
 I would really like to understand all the pieces to this scenario.

 The debug I posted was from the HQ router/GK/CUBE, I'm not sure how to
 read it yet, so I can't say what call leg it represents.

 We are talking about the same scenario you mention in your reply.

 CUCM  via GK-0Trunk (No MTP configured) (Wait for H245 unchecked)
 GK configured with a Remote Zone using a outvia to local CUBE.
 CUBE is configured with One inbound dial-peer 011! with a fixed codec
 of g.711 and an outbound dial-peer targeting RAS (default g.729)
 The call setup works and I can answer the call but the rtp never works.

 Please confirm my understanding of the problem.
 1.) CUCM does ARQ/setup via GK
 2.) GK sends LRQ to BBGK and gets LCF that it's a routable DN
 3.) GK tells CUCM to target CUBE for H.323 call setup because of Outvia
 config for BBGK Zone.
 4.) CUCM sends H.225 to CUBE which triggers CUBE to do H.225 to
 endpoint in BBGK Zone.
 5.) CUBE waits for h.245 TCS and doesn't send H.225 connect back to
 CUCM
 6.) BBGK Endpoint doesn't send any TCS and causes CUBE to wait /timeout
 for H.245
 7.) Call fails with CUBE disconnecting both BBGK call leg and CUCM call
 leg.

 I think this is due to the fact that CUCM (without MTP) is forced to do
 slow start , while CUBE will automatically do fast start.
 As I understand , CUBE can't compensate for the difference between
 slow/fast start call legs.

 So is the only option to have an MTP configured at CUCM side?
 Can the CUBE be forced to do slow start? Would that fix the issue?




 On Fri, Dec 2, 2011 at 1:20 AM, Ashraf Ayyash wrote:

> 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 and you have CUBE with Transcoder in it and the
> issue you trying to trace is that once you connected the call it got
> dropped by the remote GK ?
>
> On the CUBE you have inbound dial-peer with codec G711 and outbound
> dial-peer with G729 and then you have transcoder to fix this in the
> cube , but on the remote GK you have dial-peer with G711u call only,
>
> if any of the above is not what you have please correct me ,
>
>
>
> Ash
>
> On Thu, Dec 1, 2011 at 2:53 PM, ccielabrat 
> wrote:
> > 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?
> >

[OSL | CCIE_Voice] Cisco 2811

[Errol Abrahams]

Hi All,

I have a Cisco 2811 router with a VWIC-2MFT-T1 card and during bootup time,
only the one channel T1 0/1/0 becomes active. The other channel which is T1
0/1/1 doesn't respond, but yet the 'AL' led on the card stays on. If I shut
down the controller T1 0/1/0 then this channel responds but not the other
one. Is there a command on the Cisco 2811 to active the other channel.

I would appreciate your responds regarding this matter..thnx.

Cheers

Errol
___
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Re: [OSL | CCIE_Voice] Need some help with GK/CUBE Remote Zone call debugging.

[ccielabrat]

Hi Chris,
So is CUBE waiting for CUCM to send TCS ?
And holding up the whole process?

Also, I guess this means the the BBGK is waiting for TCS from CUBE.
Is CUBE sending TCS independent of the CUCM call leg? or does it wait for
CUCM TCS before sending TCS to BBGK?



On Fri, Dec 2, 2011 at 3:08 PM, Chris Martin  wrote:

> Because the IOS gateway trunks follow the rfc that the initiator will send
> TCS, while by default CUCM waits for the TCS from the end point.  Per the
> SRND this was done for potential video issues, so either you may have video
> problems with certain endpoints or voice issues with gateways trunks.  You
> can review this on CUCM SRND, search for TCS.
>
> HTH,
> Chris
>
>
> On Fri, Dec 2, 2011 at 12:45 PM, datucha123 datucha123 <
> datucha...@gmail.com> wrote:
>
>> By the way, why the BBGK endpoint does not send the TCS?
>>
>>
>>
>>
>> On Fri, Dec 2, 2011 at 10:02 PM, ccielabrat  wrote:
>>
>>> Hi Ash,
>>>
>>> Thanks very much for taking the time to reply.
>>> I would really like to understand all the pieces to this scenario.
>>>
>>> The debug I posted was from the HQ router/GK/CUBE, I'm not sure how to
>>> read it yet, so I can't say what call leg it represents.
>>>
>>> We are talking about the same scenario you mention in your reply.
>>>
>>> CUCM  via GK-0Trunk (No MTP configured) (Wait for H245 unchecked)
>>> GK configured with a Remote Zone using a outvia to local CUBE.
>>> CUBE is configured with One inbound dial-peer 011! with a fixed codec of
>>> g.711 and an outbound dial-peer targeting RAS (default g.729)
>>> The call setup works and I can answer the call but the rtp never works.
>>>
>>> Please confirm my understanding of the problem.
>>> 1.) CUCM does ARQ/setup via GK
>>> 2.) GK sends LRQ to BBGK and gets LCF that it's a routable DN
>>> 3.) GK tells CUCM to target CUBE for H.323 call setup because of Outvia
>>> config for BBGK Zone.
>>> 4.) CUCM sends H.225 to CUBE which triggers CUBE to do H.225 to endpoint
>>> in BBGK Zone.
>>> 5.) CUBE waits for h.245 TCS and doesn't send H.225 connect back to CUCM
>>> 6.) BBGK Endpoint doesn't send any TCS and causes CUBE to wait /timeout
>>> for H.245
>>> 7.) Call fails with CUBE disconnecting both BBGK call leg and CUCM call
>>> leg.
>>>
>>> I think this is due to the fact that CUCM (without MTP) is forced to do
>>> slow start , while CUBE will automatically do fast start.
>>> As I understand , CUBE can't compensate for the difference between
>>> slow/fast start call legs.
>>>
>>> So is the only option to have an MTP configured at CUCM side?
>>> Can the CUBE be forced to do slow start? Would that fix the issue?
>>>
>>>
>>>
>>>
>>> On Fri, Dec 2, 2011 at 1:20 AM, Ashraf Ayyash wrote:
>>>
 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 and you have CUBE with Transcoder in it and the
 issue you trying to trace is that once you connected the call it got
 dropped by the remote GK ?

 On the CUBE you have inbound dial-peer with codec G711 and outbound
 dial-peer with G729 and then you have transcoder to fix this in the
 cube , but on the remote GK you have dial-peer with G711u call only,

 if any of the above is not what you have please correct me ,



 Ash

 On Thu, Dec 1, 2011 at 2:53 PM, ccielabrat 
 wrote:
 > 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 allocate the xcoder to get around the
 codec
 > mismatch.
 >
 > Output from Debug of H245 ASN1 on HQ/GK/CUBE
 >
 > Dec  1 20:45:36.806: h245_decode_one_pdu: more_pdus = 0,
 bytesLeftToDecode =
 > 97
 > Dec  1 20:45:36.806: H245 MSC INCOMING ENCODE BUFFER::=
 >
 0270010600088175000A801380003C000101010CC0010001000680240001058124080105822280058322C005848501408585011080002B85015000820300010002000301000400052B
 > Dec  1 20:45:36.806:
 > Dec  1 20:45:36.806: H245 MSC INCOMING PDU ::=
 >
 > value MultimediaSystemControlMessage ::= request :
 terminalCapabilitySet :
 > {
 >   sequenceNumber 1
 >   protocolIdentifier { 0 0 8 245 0 10 }
 >   multiplexCapability h2250Capability :
 >   {
 > maximumAudioDelayJitter 60
 > receiveMultipointCapability
 > {
 >   multicastCapability FALSE
 >   multiUniCastConference FALSE
 >   mediaDistribu

Re: [OSL | CCIE_Voice] DSPFarms for SRST

[Gurpreet Singh Kukreja]

Hi,

You can configure the dspfarms under telephony-service as far as i know but
not under call-m-f. Under telephony service, you can use the sdsfarm
commands to do so.

- Gurpreet


On Fri, Dec 2, 2011 at 11:22 AM, datucha123 datucha123  wrote:

> Hello,
>
> I can configure the Transcoding for CUCME SRST.
>
> But as I guess the call-manager-fallback does not support registering the
> dspfarm profiles. So no Transcoder or any other profile is supported in
> Traditional SRST.
>
> Am I right?
>
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> visit www.ipexpert.com
>
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> www.PlatinumPlacement.com
>
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Re: [OSL | CCIE_Voice] Need some help with GK/CUBE Remote Zone call debugging.

[Chris Martin]

Because the IOS gateway trunks follow the rfc that the initiator will send
TCS, while by default CUCM waits for the TCS from the end point.  Per the
SRND this was done for potential video issues, so either you may have video
problems with certain endpoints or voice issues with gateways trunks.  You
can review this on CUCM SRND, search for TCS.

HTH,
Chris

On Fri, Dec 2, 2011 at 12:45 PM, datucha123 datucha123  wrote:

> By the way, why the BBGK endpoint does not send the TCS?
>
>
>
>
> On Fri, Dec 2, 2011 at 10:02 PM, ccielabrat  wrote:
>
>> Hi Ash,
>>
>> Thanks very much for taking the time to reply.
>> I would really like to understand all the pieces to this scenario.
>>
>> The debug I posted was from the HQ router/GK/CUBE, I'm not sure how to
>> read it yet, so I can't say what call leg it represents.
>>
>> We are talking about the same scenario you mention in your reply.
>>
>> CUCM  via GK-0Trunk (No MTP configured) (Wait for H245 unchecked)
>> GK configured with a Remote Zone using a outvia to local CUBE.
>> CUBE is configured with One inbound dial-peer 011! with a fixed codec of
>> g.711 and an outbound dial-peer targeting RAS (default g.729)
>> The call setup works and I can answer the call but the rtp never works.
>>
>> Please confirm my understanding of the problem.
>> 1.) CUCM does ARQ/setup via GK
>> 2.) GK sends LRQ to BBGK and gets LCF that it's a routable DN
>> 3.) GK tells CUCM to target CUBE for H.323 call setup because of Outvia
>> config for BBGK Zone.
>> 4.) CUCM sends H.225 to CUBE which triggers CUBE to do H.225 to endpoint
>> in BBGK Zone.
>> 5.) CUBE waits for h.245 TCS and doesn't send H.225 connect back to CUCM
>> 6.) BBGK Endpoint doesn't send any TCS and causes CUBE to wait /timeout
>> for H.245
>> 7.) Call fails with CUBE disconnecting both BBGK call leg and CUCM call
>> leg.
>>
>> I think this is due to the fact that CUCM (without MTP) is forced to do
>> slow start , while CUBE will automatically do fast start.
>> As I understand , CUBE can't compensate for the difference between
>> slow/fast start call legs.
>>
>> So is the only option to have an MTP configured at CUCM side?
>> Can the CUBE be forced to do slow start? Would that fix the issue?
>>
>>
>>
>>
>> On Fri, Dec 2, 2011 at 1:20 AM, Ashraf Ayyash wrote:
>>
>>> 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 and you have CUBE with Transcoder in it and the
>>> issue you trying to trace is that once you connected the call it got
>>> dropped by the remote GK ?
>>>
>>> On the CUBE you have inbound dial-peer with codec G711 and outbound
>>> dial-peer with G729 and then you have transcoder to fix this in the
>>> cube , but on the remote GK you have dial-peer with G711u call only,
>>>
>>> if any of the above is not what you have please correct me ,
>>>
>>>
>>>
>>> Ash
>>>
>>> On Thu, Dec 1, 2011 at 2:53 PM, ccielabrat  wrote:
>>> > 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 allocate the xcoder to get around the
>>> codec
>>> > mismatch.
>>> >
>>> > Output from Debug of H245 ASN1 on HQ/GK/CUBE
>>> >
>>> > Dec  1 20:45:36.806: h245_decode_one_pdu: more_pdus = 0,
>>> bytesLeftToDecode =
>>> > 97
>>> > Dec  1 20:45:36.806: H245 MSC INCOMING ENCODE BUFFER::=
>>> >
>>> 0270010600088175000A801380003C000101010CC0010001000680240001058124080105822280058322C005848501408585011080002B85015000820300010002000301000400052B
>>> > Dec  1 20:45:36.806:
>>> > Dec  1 20:45:36.806: H245 MSC INCOMING PDU ::=
>>> >
>>> > value MultimediaSystemControlMessage ::= request :
>>> terminalCapabilitySet :
>>> > {
>>> >   sequenceNumber 1
>>> >   protocolIdentifier { 0 0 8 245 0 10 }
>>> >   multiplexCapability h2250Capability :
>>> >   {
>>> > maximumAudioDelayJitter 60
>>> > receiveMultipointCapability
>>> > {
>>> >   multicastCapability FALSE
>>> >   multiUniCastConference FALSE
>>> >   mediaDistributionCapability
>>> >   {
>>> >
>>> > {
>>> >   centralizedControl FALSE
>>> >   distributedControl FALSE
>>> >   centralizedAudio FALSE
>>> >   distributedAudio FA
>>> > HQ#LSE
>>> >   centralizedVideo FALSE
>>> >   distributedVideo FALSE
>>> > }
>>> >   }
>>> > }
>>> > transmitMultipointCapability
>>> > {
>>> >   multicastCapability FALSE
>

[OSL | CCIE_Voice] Automated Reply Re: "CCIE_Voice Digest, Vol 70, Issue 14"

[stewart . mcfarlane]

This is an automated reply to your message "CCIE_Voice Digest, Vol 70, Issue 
14" 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 week (w/c 17/10/11).

If you require any assistance please contact the provista office on 
08456424642.  For technical assistance please contact 
serviced...@provista-uk.com.

Thanks

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[OSL | CCIE_Voice] SRST MoH issue

[datucha123 datucha123]

Hi Guys,
In SRST mode Branch 1 , I cannot hear the MOH from Flash ,but I hear beep
sound. There is no MOH playing for internal calls from BR1 phn 1 to BR1 Phn
2.
But when I put the PSTN caller on hold the PSTN phn can hear the MOH. I
tried the debug ephone MOHit says:

*No MOH entry for DN 1*

*Dec  2 19:27:37.678: ifs_read flash:MobileConnectOn.ulaw.wav end of file
at 15078 read 6832 = 21910
*Dec  2 19:27:37.682: moh tail fill from 46 at 0x66BBACF4 length 1168
*Dec  2 19:27:38.254: ephone_hold_resume ignored for s2s set on dn=1 chan=1
hold=1 callID=1547
**Dec  2 19:27:39.922: No MOH entry for DN 1
**Dec  2 19:27:39.926: ephone_hold_resume ignored for s2s set on dn=1
chan=1 hold=0 callID=1547
*Dec  2 19:27:40.678: ifs_read flash:MobileConnectOn.ulaw.wav end of file
at 17214 read 4696 = 21910
*Dec  2 19:27:40.678: moh tail fill from 46 at 0x66BB855C length 3304
*Dec  2 19:27:40.882: MoH route If Vlan11 ETHERNET 177.2.11.1 via ARP
*Dec  2 19:27:40.882: MoH route If Loopback0 46 177.1.254.2 via 177.1.254.2
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Re: [OSL | CCIE_Voice] Need some help with GK/CUBE Remote Zone call debugging.

[datucha123 datucha123]

By the way, why the BBGK endpoint does not send the TCS?




On Fri, Dec 2, 2011 at 10:02 PM, ccielabrat  wrote:

> Hi Ash,
>
> Thanks very much for taking the time to reply.
> I would really like to understand all the pieces to this scenario.
>
> The debug I posted was from the HQ router/GK/CUBE, I'm not sure how to
> read it yet, so I can't say what call leg it represents.
>
> We are talking about the same scenario you mention in your reply.
>
> CUCM  via GK-0Trunk (No MTP configured) (Wait for H245 unchecked)
> GK configured with a Remote Zone using a outvia to local CUBE.
> CUBE is configured with One inbound dial-peer 011! with a fixed codec of
> g.711 and an outbound dial-peer targeting RAS (default g.729)
> The call setup works and I can answer the call but the rtp never works.
>
> Please confirm my understanding of the problem.
> 1.) CUCM does ARQ/setup via GK
> 2.) GK sends LRQ to BBGK and gets LCF that it's a routable DN
> 3.) GK tells CUCM to target CUBE for H.323 call setup because of Outvia
> config for BBGK Zone.
> 4.) CUCM sends H.225 to CUBE which triggers CUBE to do H.225 to endpoint
> in BBGK Zone.
> 5.) CUBE waits for h.245 TCS and doesn't send H.225 connect back to CUCM
> 6.) BBGK Endpoint doesn't send any TCS and causes CUBE to wait /timeout
> for H.245
> 7.) Call fails with CUBE disconnecting both BBGK call leg and CUCM call
> leg.
>
> I think this is due to the fact that CUCM (without MTP) is forced to do
> slow start , while CUBE will automatically do fast start.
> As I understand , CUBE can't compensate for the difference between
> slow/fast start call legs.
>
> So is the only option to have an MTP configured at CUCM side?
> Can the CUBE be forced to do slow start? Would that fix the issue?
>
>
>
>
> On Fri, Dec 2, 2011 at 1:20 AM, Ashraf Ayyash wrote:
>
>> 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 and you have CUBE with Transcoder in it and the
>> issue you trying to trace is that once you connected the call it got
>> dropped by the remote GK ?
>>
>> On the CUBE you have inbound dial-peer with codec G711 and outbound
>> dial-peer with G729 and then you have transcoder to fix this in the
>> cube , but on the remote GK you have dial-peer with G711u call only,
>>
>> if any of the above is not what you have please correct me ,
>>
>>
>>
>> Ash
>>
>> On Thu, Dec 1, 2011 at 2:53 PM, ccielabrat  wrote:
>> > 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 allocate the xcoder to get around the
>> codec
>> > mismatch.
>> >
>> > Output from Debug of H245 ASN1 on HQ/GK/CUBE
>> >
>> > Dec  1 20:45:36.806: h245_decode_one_pdu: more_pdus = 0,
>> bytesLeftToDecode =
>> > 97
>> > Dec  1 20:45:36.806: H245 MSC INCOMING ENCODE BUFFER::=
>> >
>> 0270010600088175000A801380003C000101010CC0010001000680240001058124080105822280058322C005848501408585011080002B85015000820300010002000301000400052B
>> > Dec  1 20:45:36.806:
>> > Dec  1 20:45:36.806: H245 MSC INCOMING PDU ::=
>> >
>> > value MultimediaSystemControlMessage ::= request :
>> terminalCapabilitySet :
>> > {
>> >   sequenceNumber 1
>> >   protocolIdentifier { 0 0 8 245 0 10 }
>> >   multiplexCapability h2250Capability :
>> >   {
>> > maximumAudioDelayJitter 60
>> > receiveMultipointCapability
>> > {
>> >   multicastCapability FALSE
>> >   multiUniCastConference FALSE
>> >   mediaDistributionCapability
>> >   {
>> >
>> > {
>> >   centralizedControl FALSE
>> >   distributedControl FALSE
>> >   centralizedAudio FALSE
>> >   distributedAudio FA
>> > HQ#LSE
>> >   centralizedVideo FALSE
>> >   distributedVideo FALSE
>> > }
>> >   }
>> > }
>> > transmitMultipointCapability
>> > {
>> >   multicastCapability FALSE
>> >   multiUniCastConference FALSE
>> >   mediaDistributionCapability
>> >   {
>> >
>> > {
>> >   centralizedControl FALSE
>> >   distributedControl FALSE
>> >   centralizedAudio FALSE
>> >   distributedAudio FALSE
>> >   centralizedVideo FALSE
>> >   distributedVideo FALSE
>> > }
>> >   }
>> > }
>> > receiveAndTransmitMultipointCapability
>> > {
>> >   multicastCapability FALSE
>> >   

Re: [OSL | CCIE_Voice] Need some help with GK/CUBE Remote Zone call debugging.

[datucha123 datucha123]

You can uncheck the "wait for far end H245 Terminal Capability Set" in the
H225 Trunk configuration, so that CUCM will not go to Timeout for H245.

On Fri, Dec 2, 2011 at 10:02 PM, ccielabrat  wrote:

> Hi Ash,
>
> Thanks very much for taking the time to reply.
> I would really like to understand all the pieces to this scenario.
>
> The debug I posted was from the HQ router/GK/CUBE, I'm not sure how to
> read it yet, so I can't say what call leg it represents.
>
> We are talking about the same scenario you mention in your reply.
>
> CUCM  via GK-0Trunk (No MTP configured) (Wait for H245 unchecked)
> GK configured with a Remote Zone using a outvia to local CUBE.
> CUBE is configured with One inbound dial-peer 011! with a fixed codec of
> g.711 and an outbound dial-peer targeting RAS (default g.729)
> The call setup works and I can answer the call but the rtp never works.
>
> Please confirm my understanding of the problem.
> 1.) CUCM does ARQ/setup via GK
> 2.) GK sends LRQ to BBGK and gets LCF that it's a routable DN
> 3.) GK tells CUCM to target CUBE for H.323 call setup because of Outvia
> config for BBGK Zone.
> 4.) CUCM sends H.225 to CUBE which triggers CUBE to do H.225 to endpoint
> in BBGK Zone.
> 5.) CUBE waits for h.245 TCS and doesn't send H.225 connect back to CUCM
> 6.) BBGK Endpoint doesn't send any TCS and causes CUBE to wait /timeout
> for H.245
> 7.) Call fails with CUBE disconnecting both BBGK call leg and CUCM call
> leg.
>
> I think this is due to the fact that CUCM (without MTP) is forced to do
> slow start , while CUBE will automatically do fast start.
> As I understand , CUBE can't compensate for the difference between
> slow/fast start call legs.
>
> So is the only option to have an MTP configured at CUCM side?
> Can the CUBE be forced to do slow start? Would that fix the issue?
>
>
>
>
> On Fri, Dec 2, 2011 at 1:20 AM, Ashraf Ayyash wrote:
>
>> 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 and you have CUBE with Transcoder in it and the
>> issue you trying to trace is that once you connected the call it got
>> dropped by the remote GK ?
>>
>> On the CUBE you have inbound dial-peer with codec G711 and outbound
>> dial-peer with G729 and then you have transcoder to fix this in the
>> cube , but on the remote GK you have dial-peer with G711u call only,
>>
>> if any of the above is not what you have please correct me ,
>>
>>
>>
>> Ash
>>
>> On Thu, Dec 1, 2011 at 2:53 PM, ccielabrat  wrote:
>> > 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 allocate the xcoder to get around the
>> codec
>> > mismatch.
>> >
>> > Output from Debug of H245 ASN1 on HQ/GK/CUBE
>> >
>> > Dec  1 20:45:36.806: h245_decode_one_pdu: more_pdus = 0,
>> bytesLeftToDecode =
>> > 97
>> > Dec  1 20:45:36.806: H245 MSC INCOMING ENCODE BUFFER::=
>> >
>> 0270010600088175000A801380003C000101010CC0010001000680240001058124080105822280058322C005848501408585011080002B85015000820300010002000301000400052B
>> > Dec  1 20:45:36.806:
>> > Dec  1 20:45:36.806: H245 MSC INCOMING PDU ::=
>> >
>> > value MultimediaSystemControlMessage ::= request :
>> terminalCapabilitySet :
>> > {
>> >   sequenceNumber 1
>> >   protocolIdentifier { 0 0 8 245 0 10 }
>> >   multiplexCapability h2250Capability :
>> >   {
>> > maximumAudioDelayJitter 60
>> > receiveMultipointCapability
>> > {
>> >   multicastCapability FALSE
>> >   multiUniCastConference FALSE
>> >   mediaDistributionCapability
>> >   {
>> >
>> > {
>> >   centralizedControl FALSE
>> >   distributedControl FALSE
>> >   centralizedAudio FALSE
>> >   distributedAudio FA
>> > HQ#LSE
>> >   centralizedVideo FALSE
>> >   distributedVideo FALSE
>> > }
>> >   }
>> > }
>> > transmitMultipointCapability
>> > {
>> >   multicastCapability FALSE
>> >   multiUniCastConference FALSE
>> >   mediaDistributionCapability
>> >   {
>> >
>> > {
>> >   centralizedControl FALSE
>> >   distributedControl FALSE
>> >   centralizedAudio FALSE
>> >   distributedAudio FALSE
>> >   centralizedVideo FALSE
>> >   distributedVideo FALSE
>> > }
>> >   }
>> > }
>> > receiveAndTrans

Re: [OSL | CCIE_Voice] Need some help with GK/CUBE Remote Zone call debugging.

[ccielabrat]

Hi Ash,

Thanks very much for taking the time to reply.
I would really like to understand all the pieces to this scenario.

The debug I posted was from the HQ router/GK/CUBE, I'm not sure how to read
it yet, so I can't say what call leg it represents.

We are talking about the same scenario you mention in your reply.

CUCM  via GK-0Trunk (No MTP configured) (Wait for H245 unchecked)
GK configured with a Remote Zone using a outvia to local CUBE.
CUBE is configured with One inbound dial-peer 011! with a fixed codec of
g.711 and an outbound dial-peer targeting RAS (default g.729)
The call setup works and I can answer the call but the rtp never works.

Please confirm my understanding of the problem.
1.) CUCM does ARQ/setup via GK
2.) GK sends LRQ to BBGK and gets LCF that it's a routable DN
3.) GK tells CUCM to target CUBE for H.323 call setup because of Outvia
config for BBGK Zone.
4.) CUCM sends H.225 to CUBE which triggers CUBE to do H.225 to endpoint in
BBGK Zone.
5.) CUBE waits for h.245 TCS and doesn't send H.225 connect back to CUCM
6.) BBGK Endpoint doesn't send any TCS and causes CUBE to wait /timeout for
H.245
7.) Call fails with CUBE disconnecting both BBGK call leg and CUCM call leg.

I think this is due to the fact that CUCM (without MTP) is forced to do
slow start , while CUBE will automatically do fast start.
As I understand , CUBE can't compensate for the difference between
slow/fast start call legs.

So is the only option to have an MTP configured at CUCM side?
Can the CUBE be forced to do slow start? Would that fix the issue?



On Fri, Dec 2, 2011 at 1:20 AM, Ashraf Ayyash  wrote:

> 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 and you have CUBE with Transcoder in it and the
> issue you trying to trace is that once you connected the call it got
> dropped by the remote GK ?
>
> On the CUBE you have inbound dial-peer with codec G711 and outbound
> dial-peer with G729 and then you have transcoder to fix this in the
> cube , but on the remote GK you have dial-peer with G711u call only,
>
> if any of the above is not what you have please correct me ,
>
>
>
> Ash
>
> On Thu, Dec 1, 2011 at 2:53 PM, ccielabrat  wrote:
> > 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 allocate the xcoder to get around the codec
> > mismatch.
> >
> > Output from Debug of H245 ASN1 on HQ/GK/CUBE
> >
> > Dec  1 20:45:36.806: h245_decode_one_pdu: more_pdus = 0,
> bytesLeftToDecode =
> > 97
> > Dec  1 20:45:36.806: H245 MSC INCOMING ENCODE BUFFER::=
> >
> 0270010600088175000A801380003C000101010CC0010001000680240001058124080105822280058322C005848501408585011080002B85015000820300010002000301000400052B
> > Dec  1 20:45:36.806:
> > Dec  1 20:45:36.806: H245 MSC INCOMING PDU ::=
> >
> > value MultimediaSystemControlMessage ::= request : terminalCapabilitySet
> :
> > {
> >   sequenceNumber 1
> >   protocolIdentifier { 0 0 8 245 0 10 }
> >   multiplexCapability h2250Capability :
> >   {
> > maximumAudioDelayJitter 60
> > receiveMultipointCapability
> > {
> >   multicastCapability FALSE
> >   multiUniCastConference FALSE
> >   mediaDistributionCapability
> >   {
> >
> > {
> >   centralizedControl FALSE
> >   distributedControl FALSE
> >   centralizedAudio FALSE
> >   distributedAudio FA
> > HQ#LSE
> >   centralizedVideo FALSE
> >   distributedVideo FALSE
> > }
> >   }
> > }
> > transmitMultipointCapability
> > {
> >   multicastCapability FALSE
> >   multiUniCastConference FALSE
> >   mediaDistributionCapability
> >   {
> >
> > {
> >   centralizedControl FALSE
> >   distributedControl FALSE
> >   centralizedAudio FALSE
> >   distributedAudio FALSE
> >   centralizedVideo FALSE
> >   distributedVideo FALSE
> > }
> >   }
> > }
> > receiveAndTransmitMultipointCapability
> > {
> >   multicastCapability FALSE
> >   multiUniCastConference FALSE
> >   mediaDistributionCapability
> >   {
> >
> > {
> >   centralizedControl FALSE
> >   distributedControl FALSE
> >   centralizedAudio FALSE
> >   distributedAudio FA

[OSL | CCIE_Voice] DSPFarms for SRST

[datucha123 datucha123]

Hello,

I can configure the Transcoding for CUCME SRST.

But as I guess the call-manager-fallback does not support registering the
dspfarm profiles. So no Transcoder or any other profile is supported in
Traditional SRST.

Am I right?
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[OSL | CCIE_Voice] Notify DTMF for CUCM

[datucha123 datucha123]

I know, that for CUCM to receive the SIP-Notify DTMFs, I have to enable the
"Accept Unsolicited Notification" in the SIP Trunk Security Profile.

But I cannot force CUCM to negotiate the SIP-Notify for Outgoing calls.

When I configure the SIP-Notify DTMF method for dial-peer pointing to CUCM
(Also this dial-peer is matched when the CUCM calls the GW), the "show
sip-ua calls" command shows that the Inband DTMF is negotiated, while I
want the SIP Notify.
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[OSL | CCIE_Voice] RFC2833 with MTP

[datucha123 datucha123]

Hello,

I have configured the Routers Software MTP for G729 codec with SIP Trunk.
I am using RFC2833 (RTP-NTE) DTMF type on the gateway (CUBE).
When I configure the "MTP Required" on the SIP Trunk, the DTMFs are not
recognized by the UCCX, the call goes directly to operator (UCCX is using
default AA Script).

But when I remove the "MTP Required" on the SIP Trunk, so that the calls
will be allocated MTP dynamically, the DTMF works fine (UCCX gets the DTMF
tones normally).
I have also noticed that when the MTP Required is configured on a SIP
Trunk, the Routers MTP is allocation/Used and RFC2833 DTMFs are not
recognized by the UCCX.
But when I remove the MTP Required from the SIP Trunk, the Transcoder is
allocated/used dynamically (UCCX does not support RFC2833), and the RFC2833
DTMFs are successfully translated to out of band and UCCX understand them
normally.

Can anybody explain why is it so?
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[OSL | CCIE_Voice] DTMF issues (Inband to Out of band)

[datucha123 datucha123]

Hello,

As I know, the command "dtmf-relay rtp-nte digit-drop" when using RFC2833
on one side of CUBE, and Out of Band DTMF types on the other side.

But somehow that does not work in my case. Here are my CUBE dial-peer that
are configured:

dial-peer voice 10 voip
 description to CUCM
 translation-profile incoming 335
 destination-pattern ^[1247]...$
 session protocol sipv2
 session target ipv4:172.16.4.121
 incoming called-number [358]...
 dtmf-relay sip-kpml
 no vad
!
dial-peer voice 11 voip
 description to CUCME
 destination-pattern ^[358]...
 voice-class sip dtmf-relay force rtp-nte
 voice-class sip early-offer forced
 session protocol sipv2
 session target ipv4:177.1.254.3
 incoming called-number ^[1247]...$
 dtmf-relay rtp-nte digit-drop
 no vad

So when the CUCME calls the CUCM (UCCX) the DTMF are not recognized by the
UCCX. So I guess the "dtmf-relay rtp-nte digit-drop" command does not work.
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Re: [OSL | CCIE_Voice] Need Help with MoH

[datucha123 datucha123]

I have tested it in my Lab, the the MoH was working without MTP either
through Xcoder or G729 enabled in IPVMS service.

So that BUG does not effect my CUCM

On Thu, Dec 1, 2011 at 10:13 PM, datucha123 datucha123  wrote:

> 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 the MoH was transcoded :)
>
> But the BUG you have sent, is very intestring, so I will also test it
> tomorrow.
>
>
> On Thu, Dec 1, 2011 at 7:57 PM, Mohammed Al Baqari <
> baqari.voic...@gmail.com> wrote:
>
>>  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   
>>
>> G0 L 85   N   ORG T4 g729r8  VOIPP
>> 142.2.64.254:17672   
>>
>> G0 L 86   N   ORG T4 g729r8  VOIPP
>> 142.2.64.254:17502
>>
>> G0 L 88   N   ORG T4 g711ulawVOIPP
>> 0.0.0.0:0
>>
>> ** **
>>
>> The interesting part is this. It wasn’t working without MTP (either I am
>> using g711 on MOH which needs XCODE or even if I change MOH to G729 which
>> doesn’t require XCODE). Finally I found that I am hitting this bug. So it’s
>> a must to use MTP over SIP trunk for MOH to work either with XCODE or not.
>> 
>>
>> ** **
>>
>> *CSCso85618 Bug Details *
>>
>> ** **
>>
>> Top of Form
>>
>> *No Audio when Call is put on hold by remote party over Sip Trunk *
>>
>> *Symptoms*:
>>
>> 1. The MOH does not work between CM and CME.
>>
>> 2. There is no audio on the CME endpoint when the remote CCM party
>> resumes the
>> call on hold, conferences, or transfers with another CCM endpoint
>> (scenario:
>> CME - CUBE - CCM).
>>
>> *Conditions*: Symptom 1 is observed if the phone registered to the CME
>> is put on
>> hold by the CM, then the CME phone does not hear the MOH.
>>
>> Symptom 2 is observed if the CCM endpoint does a conference, hold, or
>> transfer.
>>
>> Workaround: Use an MTP.
>>
>> Regards,
>>
>> Mohammed Al Baqari
>>
>> ** **
>>
>> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
>> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *datucha123
>> datucha123
>> *Sent:* Wednesday, November 30, 2011 10:59 PM
>> *To:* ccie_voice@onlinestudylist.com
>> *Subject:* [OSL | CCIE_Voice] Need Help with MoH
>>
>> ** **
>>
>> Hello,
>>
>>  
>>
>> I am facing a Problem with MoH over the SIP/H323 Trunk with MTP.
>>
>>  
>>
>> I know, that the multicast MoH is not supported when the MTP is check in
>> the H323 gateway, but I have a problem with Unicast MoH with transcoding.
>> 
>>
>>  
>>
>> But I have the following configuration.
>>
>>  
>>
>> Only G711U is activated for MoH (default). 
>>
>>  
>>
>> I have a SIP Trunk towards CUCME, this SIP Trunk is in G729 Region, so
>> that this Region is using G729 to CUCM IP Phones, and to MoH regions as
>> well:
>>
>>  
>>
>> So I have 
>>
>> HQ_Region  -  G711 Inside and G729 to SIP_Trunk_Region. 
>>
>> SIP_Trunk_Region  - G711 inside and G729 to HQ_Region.
>>
>>  
>>
>> So the SIP Trunk is assigned the SIP_Trunk_Region (Through DP) and the IP
>> Phones along with the MoH Server are assigned the HQ_Region. 
>>
>>  
>>
>> Also I have an G729 MTP on a Router, which is assigned the MTP_Region
>> (Which is using G729 to all other Regions).
>>
>> Transcoder is assigne the HQ_Region.
>>
>>  
>>
>> So when I call the SIP Trunk (CUCME IP Phones) the call is successful and
>> negotiated codec is G729, and MTP is also allocated, as I have checked the
>> MTP Required Checkbox. 
>>
>>  
>>
>> But when I press the Hold key on the CUCM IP Phone, the CUCME IP Phone
>> hears a Beeps instead of MoH Music. 
>>
>>  
>>
>> I cannot get how to make transcoder to transcode the MoH from G711 to
>> G729, so that the MTP will be also used on the SIP Trunk.
>>
>>  
>>
>> BTW, when I remove the MTP Required from the SIP Trunk, the MoH is
>> transcoded, and the remote IP Phone (CUCME) can hear the music. 
>>
>>  
>>
>> But I want to have an MTP checked as well, and in this case the MoH music
>> does not play. (Only Beeps).
>>
>>  
>>
>> Please help me.
>>
>>  
>>
>>  
>>
>>  
>>
>
>
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Re: [OSL | CCIE_Voice] Introducing myself

[Nicolas MICHEL]

Hey Michael.

Thanks again for all the help provided with the CCIE RS when I was studying
for it :)

How far from the CCIE Voice are you now ?

I m just starting, building a phones lab and then I'll be using IPX and
some other vendors as well I guess 
I remember you were building a rack with friends, how far are you from
there ? :)

Seeing that this Mailing list is far more active than the RS one ! Cheers !!


Nic




2011/12/2 Michael Miller 

> Hello Nicholas,
>
> Its nice to see some familiar faces from the RS OSL boards. =)
>
> Congrats on passing the RS, and good luck on the Voice!
>
> Thanks,
>
> Michael
>
> On Fri, Dec 2, 2011 at 11:30 AM, Emanuel Damasceno 
> wrote:
>
>> Welcome Michel!
>>
>> You will see this is just a nice journey as RS was for you. I am the
>> opposite, I started into Voice, and when I get my CCIE, I will start on my
>> CCNP, and CCIE RS... =)
>>
>> Welcome to the UC world. I really love it and I am hoping you will love
>> it too!
>> Best regards, brother.
>> *Emanuel Damasceno*
>>
>>
>>
>>
>>   On Fri, Dec 2, 2011 at 8:11 AM, Nicolas MICHEL 
>> wrote:
>>
>>>  Hey There guys.
>>>
>>> I'm a french network engineer mainly focused into RS but as of now I m
>>> starting to deploy UC solutions and so far so good I like it.
>>> This is why I decided to pursue my 2nd CCIE into Voice and can't wait to
>>> be there yet :)
>>>
>>> I actually finished the CCNA book and the CBT nuggets for that series
>>> and now digging into CCNP stuff.
>>>
>>> I got CVOICE, CIPT1,CIPT2, Presence, CUCM+unity, Unity books. and then I
>>> ll start to read the SRND which looks awesome.
>>>
>>> I'm also building a lab to use some remote racks.
>>>
>>> If you guys have any advices, I d be glad to hear them :P
>>>
>>> Thanks for your help and cant wait to have the knowledge to ask question
>>> and answer on the OSL :)
>>>
>>> Nic
>>>
>>> --
>>> Nicolas MICHEL
>>> Ingenieur Réseaux CCIE #29410
>>>
>>>
>>>
>>>
>>>
>>>
>>> ___
>>> 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
>>> www.PlatinumPlacement.com 
>>>
>>
>>
>> ___
>> 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
>> www.PlatinumPlacement.com 
>>
>
>


-- 
Nicolas MICHEL
Ingenieur Réseaux CCIE #29410
Tel: +33 6 08 72 75 97
___
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Re: [OSL | CCIE_Voice] [EMEA Cluster & Gateway Dial Plan Assistance Munich Germany]

[brajesh kumaR]

Hi Micheal,


Start with international ,national and mobile number calls dial plan
with following. You need to manipulate called number either in CUCM or
at gateway. For ease you can use 0T at POTS dial peer so that
international call will be send with 00 prefix and national/mobile
call with 0 prefix which also might be expected from PSTN provider.


0.00!#   International calls
0.00!International calls

0.0[2-9]!#   Germany national calls
0.0[2-9]!Germany national calls

0.01[5-7]!#   Germany Mobile Numbers
0.01[5-7]!


There are many non geographical numbers in Germany which you need to
take care of if there is any dial plan requirement.

Use following links to more about Germany dial plans and specific city
codes perfixes.

http://en.wikipedia.org/wiki/Telephone_numbers_in_Germany
http://www.howtocallabroad.com/results.php?callfrom=india&callto=germany


Regards,
Brajesh.


On Thu, Dec 1, 2011 at 7:23 PM,   wrote:
> 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 be
> greatly appreciated or if you can point me to resources ( in the right
> direction) would be immensely appreciated.  Thank you.
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> ___
> 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
> www.PlatinumPlacement.com
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Re: [OSL | CCIE_Voice] Cannot call India from HQ phone 2 (VOD 5.5)

[rvaq33]

Yup  thank you for your help. It took me half day to figure it out. 
Sent from my BlackBerry® wireless handheld

-Original Message-
From: Mohd Baqari 
Date: Fri, 2 Dec 2011 13:21:45 
To: Roy Vincent Aquino
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice]  Cannot call India from HQ phone 2 (VOD 5.5)

Oh just now I saw this mail ... Good you picked up alone :) keep it up

Regards,
Mohammed Al Baqari

Sent from my iPhone

On Dec 2, 2011, at 11:38 AM, Roy Vincent Aquino  wrote:

> 
> I found the culprit
> 
> tech-prefix must be set on the PSTN router
> 
> 
> interface FastEthernet0/1
>  description To Cloud_Switch00 f0/5
>  ip address 192.168.1.11 255.255.255.0
>  duplex full
>  speed 100
>  h323-gateway voip interface
>  h323-gateway voip id PSTN-WAN ipaddr 192.168.1.11 1719
>  h323-gateway voip h323-id pstn-gw
>  h323-gateway voip tech-prefix 2#
>  h323-gateway voip bind srcaddr 192.168.1.11
> 
> 
> gatekeeper
>  zone local PSTN-WAN nuggetlabs.com 192.168.1.11
>  zone remote HQ-RTR nuggetlabs.com 10.10.200.3 1719
>  zone remote US nuggetlabs.com 10.10.110.1 1719
>  zone prefix PSTN-WAN 34*
>  zone prefix PSTN-WAN 91*
>  gw-type-prefix 2#* default-technology
>  no shutdown
> ___
> 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 
> www.PlatinumPlacement.com

___
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Re: [OSL | CCIE_Voice] Introducing myself

[Michael Miller]

Hello Nicholas,

Its nice to see some familiar faces from the RS OSL boards. =)

Congrats on passing the RS, and good luck on the Voice!

Thanks,

Michael

On Fri, Dec 2, 2011 at 11:30 AM, Emanuel Damasceno wrote:

> Welcome Michel!
>
> You will see this is just a nice journey as RS was for you. I am the
> opposite, I started into Voice, and when I get my CCIE, I will start on my
> CCNP, and CCIE RS... =)
>
> Welcome to the UC world. I really love it and I am hoping you will love it
> too!
> Best regards, brother.
> *Emanuel Damasceno*
>
>
>
>
>   On Fri, Dec 2, 2011 at 8:11 AM, Nicolas MICHEL wrote:
>
>>  Hey There guys.
>>
>> I'm a french network engineer mainly focused into RS but as of now I m
>> starting to deploy UC solutions and so far so good I like it.
>> This is why I decided to pursue my 2nd CCIE into Voice and can't wait to
>> be there yet :)
>>
>> I actually finished the CCNA book and the CBT nuggets for that series and
>> now digging into CCNP stuff.
>>
>> I got CVOICE, CIPT1,CIPT2, Presence, CUCM+unity, Unity books. and then I
>> ll start to read the SRND which looks awesome.
>>
>> I'm also building a lab to use some remote racks.
>>
>> If you guys have any advices, I d be glad to hear them :P
>>
>> Thanks for your help and cant wait to have the knowledge to ask question
>> and answer on the OSL :)
>>
>> Nic
>>
>> --
>> Nicolas MICHEL
>> Ingenieur Réseaux CCIE #29410
>>
>>
>>
>>
>>
>>
>> ___
>> 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
>> www.PlatinumPlacement.com 
>>
>
>
> ___
> 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
> www.PlatinumPlacement.com 
>
___
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[OSL | CCIE_Voice] CCNA Voice certification preparation

[Tom Ribbens]

Hi,

I realize this is slightly off topic here, but the list for CCNA Voice
bounces. Haven't tried the CCNP Voice list, but as the archives are empty,
I assume the same problem.

I took the 640-461 ICOMM exam towards CCNA Voice a couple of weeks ago, and
unfortunately failed. I had the impression a lot of topics were not covered
in the official cert guide from Cisco Press. Are there any resources to
prepare for this exam which are better suited?

Sorry again for being slightly off topic.

Regards,

Tom
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Re: [OSL | CCIE_Voice] Introducing myself

[Emanuel Damasceno]

Welcome Michel!

You will see this is just a nice journey as RS was for you. I am the
opposite, I started into Voice, and when I get my CCIE, I will start on my
CCNP, and CCIE RS... =)

Welcome to the UC world. I really love it and I am hoping you will love it
too!
Best regards, brother.
*Emanuel Damasceno*




On Fri, Dec 2, 2011 at 8:11 AM, Nicolas MICHEL wrote:

> Hey There guys.
>
> I'm a french network engineer mainly focused into RS but as of now I m
> starting to deploy UC solutions and so far so good I like it.
> This is why I decided to pursue my 2nd CCIE into Voice and can't wait to
> be there yet :)
>
> I actually finished the CCNA book and the CBT nuggets for that series and
> now digging into CCNP stuff.
>
> I got CVOICE, CIPT1,CIPT2, Presence, CUCM+unity, Unity books. and then I
> ll start to read the SRND which looks awesome.
>
> I'm also building a lab to use some remote racks.
>
> If you guys have any advices, I d be glad to hear them :P
>
> Thanks for your help and cant wait to have the knowledge to ask question
> and answer on the OSL :)
>
> Nic
>
> --
> Nicolas MICHEL
> Ingenieur Réseaux CCIE #29410
>
>
>
>
>
>
> ___
> 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
> www.PlatinumPlacement.com
>
___
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[OSL | CCIE_Voice] Introducing myself

[Nicolas MICHEL]

Hey There guys.

I'm a french network engineer mainly focused into RS but as of now I m
starting to deploy UC solutions and so far so good I like it.
This is why I decided to pursue my 2nd CCIE into Voice and can't wait to be
there yet :)

I actually finished the CCNA book and the CBT nuggets for that series and
now digging into CCNP stuff.

I got CVOICE, CIPT1,CIPT2, Presence, CUCM+unity, Unity books. and then I ll
start to read the SRND which looks awesome.

I'm also building a lab to use some remote racks.

If you guys have any advices, I d be glad to hear them :P

Thanks for your help and cant wait to have the knowledge to ask question
and answer on the OSL :)

Nic

-- 
Nicolas MICHEL
Ingenieur Réseaux CCIE #29410
___
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Re: [OSL | CCIE_Voice] Cannot call India from HQ phone 2 (VOD 5.5)

[Mohd Baqari]

Oh just now I saw this mail ... Good you picked up alone :) keep it up

Regards,
Mohammed Al Baqari

Sent from my iPhone

On Dec 2, 2011, at 11:38 AM, Roy Vincent Aquino  wrote:

> 
> I found the culprit
> 
> tech-prefix must be set on the PSTN router
> 
> 
> interface FastEthernet0/1
>  description To Cloud_Switch00 f0/5
>  ip address 192.168.1.11 255.255.255.0
>  duplex full
>  speed 100
>  h323-gateway voip interface
>  h323-gateway voip id PSTN-WAN ipaddr 192.168.1.11 1719
>  h323-gateway voip h323-id pstn-gw
>  h323-gateway voip tech-prefix 2#
>  h323-gateway voip bind srcaddr 192.168.1.11
> 
> 
> gatekeeper
>  zone local PSTN-WAN nuggetlabs.com 192.168.1.11
>  zone remote HQ-RTR nuggetlabs.com 10.10.200.3 1719
>  zone remote US nuggetlabs.com 10.10.110.1 1719
>  zone prefix PSTN-WAN 34*
>  zone prefix PSTN-WAN 91*
>  gw-type-prefix 2#* default-technology
>  no shutdown
> ___
> 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 
> www.PlatinumPlacement.com
___
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Re: [OSL | CCIE_Voice] Cannot call India from HQ phone 2 (VOD 5.5)

[Mohd Baqari]

On your PSTN GK you dont have any tech-prefix or default tech-prefix. So when 
LRQ is recieved and lookup is completed, GK won't be able to send LCF with 
india gateway IP since its not registered with any tech-prefix database.

Regards,
Mohammed Al Baqari

Sent from my iPhone

On Dec 2, 2011, at 8:05 AM, Roy Vincent Aquino  wrote:

> 
> 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 below.
> 
> HQ router:
> gatekeeper
>  zone local US nuggetlabs.com outvia VIAZONE
>  zone local SPAIN nuggetlabs.com outvia VIAZONE
>  zone local VIAZONE nuggetlabs.com
>  zone remote PSTN-WAN nuggetlabs.com 192.168.1.11 1719
>  zone prefix SPAIN 03214
>  zone prefix US 1... gw-priority 10 gk-trunk_2
>  zone prefix US 1... gw-priority 9 gk-trunk_1
>  zone prefix SPAIN 3...
>  zone prefix US 5... gw-priority 10 gk-trunk_2
>  zone prefix US 5... gw-priority 9 gk-trunk_1
>  zone prefix PSTN-WAN 91*
>  gw-type-prefix 1#* default-technology
>  no shutdown
> 
> 
> PSTN-WAN router:
> gatekeeper
>  zone local PSTN-WAN nuggetlabs.com 192.168.1.11
>  zone remote HQ-RTR nuggetlabs.com 10.10.200.3 1719
>  zone remote US nuggetlabs.com 10.10.110.1 1719
>  zone prefix PSTN-WAN 34*
>  zone prefix PSTN-WAN 91*
>  no shutdown
> 
> 
> Here's the output of debug gatekeeper main 10
> HR router:
> Dec  2 03:26:04.228: ////GK/gk_process: got a TIMER 
> event
> 
> Dec  2 03:26:04.228: ////GK/gk_handle_timers
> 
> Dec  2 03:26:04.228: ////GK/gk_handle_timers: managed 
> timer expired 0x663AA1C8 
> 
> Dec  2 03:26:04.244: ////GK/gk_process: QUEUE_EVENT 
> (minor 0) wakeup
> HQ_Router00#
> Dec  2 03:26:08.552: ////GK/gk_process: QUEUE_EVENT 
> (minor 0) wakeup
> HQ_Router00#
> Dec  2 03:26:16.768: ////GK/gk_process: QUEUE_EVENT 
> (minor 0) wakeup
> HQ_Router00#
> Dec  2 03:26:17.972: ////GK/gk_process: QUEUE_EVENT 
> (minor 0) wakeup
> Dec  2 03:26:17.976: ////GK/gk_rassrv_arq: 
> arqp=0x68BC5784,crv=0x5, answerCall=0
> Dec  2 03:26:17.976: ////GK/gk_rassrv_sep_arq: ARQ 
> Didn't use GK_AAA_PROC
> Dec  2 03:26:17.976: //0018E6630500/0018E6630500/GK/gk_dns_query: No Name 
> servers
> Dec  2 03:26:17.976: //0018E6630500/0018E6630500/GK/rassrv_get_addrinfo: 
> (916745738932) Tech-prefix match failed.
> Dec  2 03:26:17.976: //0018E6630500/0018E6630500/GK/rassrv_get_addrinfo: 
> (916745738932) Matched zone prefix 91 and remainder 6745738932
> Dec  2 03:26:17.976: 
> ////GK/gk_rassrv_get_ingress_network: returning 
> default ingress network = 1
> Dec  2 03:26:17.976: 
> //0018E6630500/0018E6630500/GK/rassrv_arq_select_viazone: about to check the 
> source side, src_zonep=0x68A89EEC
> Dec  2 03:26:17.976: 
> //0018E6630500/0018E6630500/GK/rassrv_arq_select_viazone: matched zone is US, 
> and z_invianamelen=0
> Dec  2 03:26:17.976: 
> //0018E6630500/0018E6630500/GK/rassrv_arq_select_viazone: about to check the 
> destination side, dst_zonep=0x68A8AA64
> Dec  2 03:26:17.976: 
> //0018E6630500/0018E6630500/GK/rassrv_arq_select_viazone: matched zone is 
> PSTN-WAN, and z_outvianamelen=0
> HQ_Router00#
> Dec  2 03:26:17.976: //0018E6630500/0018E6630500/GK/rassrv_get_addrinfo: No 
> tech prefix
> 
> Dec  2 03:26:17.976: //0018E6630500/0018E6630500/GK/rassrv_get_addrinfo: 
> Alias not found
> 
> Dec  2 03:26:17.976: 
> //0018E6630500/0018E6630500/GK/rassrv_put_remote_zones_from_zone_list: zone 
> PSTN-WAN
> Dec  2 03:26:17.976: //0018E6630500/0018E6630500/GK/send_lrq: seq_lrq 1, 
> use_be 0, rzone_cnt 1
> Dec  2 03:26:17.976: //0018E6630500/0018E6630500/GK/send_lrq: lrq array index 
> 4, lap 68CBEDF4
> Dec  2 03:26:17.980: //0018E6630500/0018E6630500/GK/send_lrq: sent lrq - 
> zonecount 1
> Dec  2 03:26:18.568: ////GK/gk_process: QUEUE_EVENT 
> (minor 0) wakeup
> HQ_Router00#
> Dec  2 03:26:19.228: ////GK/gk_process: got a TIMER 
> event
> 
> Dec  2 03:26:19.228: ////GK/gk_handle_timers
> 
> Dec  2 03:26:19.228: ////GK/gk_handle_timers: managed 
> timer expired 0x663AA1C8 
> 
> HQ_Router00#
> Dec  2 03:26:20.264: ////GK/gk_process: QUEUE_EVENT 
> (minor 0) wakeup
> Dec  2 03:26:20.980: ////GK/gk_process: got a TIMER 
> event
> 
> Dec  2 03:26:20.980: ////GK/gk_handle_timers
> 
> Dec  2 03:26:20.980: ////GK/gk_handle_timers: managed 
> timer expired 0x68CBEE04 
> 
> 
> 
> PSTN router:
> Dec  2 04:01:02.906: ////GK/gk_process: QUEUE_EVENT 
> (minor 0) wakeup
>

[OSL | CCIE_Voice] Cannot call India from HQ phone 2 (VOD 5.5)

[Roy Vincent Aquino]

I found the culprit

tech-prefix must be set on the PSTN router


interface FastEthernet0/1
 description To Cloud_Switch00 f0/5
 ip address 192.168.1.11 255.255.255.0
 duplex full
 speed 100
 h323-gateway voip interface
 h323-gateway voip id PSTN-WAN ipaddr 192.168.1.11 1719
 h323-gateway voip h323-id pstn-gw
 h323-gateway voip tech-prefix 2#
 h323-gateway voip bind srcaddr 192.168.1.11


gatekeeper
 zone local PSTN-WAN nuggetlabs.com 192.168.1.11
 zone remote HQ-RTR nuggetlabs.com 10.10.200.3 1719
 zone remote US nuggetlabs.com 10.10.110.1 1719
 zone prefix PSTN-WAN 34*
 zone prefix PSTN-WAN 91*
 gw-type-prefix 2#* default-technology
 no
 shutdown
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