Re: [OSL | CCIE_Voice] srst ephone-dn

2013-07-17 Thread jainpiyush2022
Show call-manager-fallback dial-peer

This will show the ephone dn dialpeer

Regards,
Piyush Jain




-Original Message-
From: Karen Johnson karen.johnson...@yahoo.ca
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Sent: Wed, 17 Jul 2013 11:15 AM
Subject: [OSL | CCIE_Voice] srst  ephone-dn

all,
 
when i configure srst autoprovision none  and create ephone-dn 4 and 
ephone-dn 5
 
it saying : ephone-dn 4 has been allocated for srst virtual dn, and it can not 
be modified
however there is no ephone-dn configured for 4 and 5 and no ephone registered.
 
- which resource that use my dn 4 and 5  ? since this is auto provision none.
- what command to check and how to delete dn 4 and 5 so I can use it.
 
K___
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Re: [OSL | CCIE_Voice] default device pool and css of already integrated devices!!

2013-07-09 Thread jainpiyush2022
You can change its settings.. and use default device pool..

Thanks and regards,
Piyush Jain

Sent from my android device.



-Original Message-
From: Drake J jdrake...@gmail.com
To: jainpiyush2...@ymail.com
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Sent: Tue, 09 Jul 2013 9:23 PM
Subject: Re: [OSL | CCIE_Voice] default device pool and css of already 
integrated devices!!

Hello Piyush,

Thanks for your response. As far as I thought the Default settings (
default device pool) are not be changed on the callmanger  . Yes / No?




On Sun, Jul 7, 2013 at 8:50 AM, jainpiyush2...@ymail.com wrote:

 Use default device pool for HQ phones and application servers like uccx,
 uc and presence.. this is the best way to do in lab.

 Thanks and regards,
 Piyush Jain

 Sent from my android device.




 -Original Message-
 From: Drake J jdrake...@gmail.com
 To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 Sent: Sun, 07 Jul 2013 6:19 AM
 Subject: [OSL | CCIE_Voice] default device pool and css of already
 integrated devices!!

 Hi Guys,

 1) Could you clarify if we need to change the default device pool   CSS
 of the devices already integrated such as unity connection , uccx , CUPS?

 2)What would be the implications if we did make the above change?

 3) If we were to retain the default device pool of the above devices then
 do  we need to have all other sites setup to have a region relation with
 the default region as g711ulaw?

 -Drake

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Re: [OSL | CCIE_Voice] Meetme Call Handler

2013-07-09 Thread jainpiyush2022
You can use one forwarding rule and one call handler.. this works perfect for 
me.

Thanks and regards,
Piyush Jain

Sent from my android device.



-Original Message-
From: Karen Johnson karen.johnson...@yahoo.ca
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Sent: Tue, 09 Jul 2013 5:30 AM
Subject: [OSL | CCIE_Voice] Meetme Call Handler

Folks,
 
Can we achieve this with One  call handler only ?  
 Meet Me When joins meet me conference it should announce the participant name 
and when somebody dials meet me number it should ask who may I say its 
calling.  and when Meetme not avail , go to HQ1 voicemail
 
Each time I try to use one call handler, Message Setting always override the 
Transfer rule, and as result call always go to HQ1 vm without asking me for 
Who May I say is calling.
 
tks
K___
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Re: [OSL | CCIE_Voice] H323 Fast Start

2013-07-09 Thread jainpiyush2022
On site b you can use Mtp with g711u and a transcoder..

Regards,
Piyush Jain

Sent from my android device.



-Original Message-
From: Edgar Feliz ejzi...@gmail.com
To: Barrera, Hugo hugo.barr...@nexusis.com
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Sent: Tue, 09 Jul 2013 7:29 AM
Subject: Re: [OSL | CCIE_Voice] H323 Fast Start

I may be wrong but I do not recall g729 being a requirement. for that task,
they are not asking HQ  SB, which would be inter-region, just F/S between
CM  R2 = SB phones dial PSTN.


Edgar


On Mon, Jul 8, 2013 at 8:03 PM, Barrera, Hugo hugo.barr...@nexusis.comwrote:

  Guy’s,

 ** **

 I used g711 for the codec, on mtp resource, but a peer of mine is stating
 I should have used g729 and by using this I would have to add a transcoder
 on the GW as well. Any thoughts? 

 ** **

 *Hugo *

 ** **

 ___
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 visit www.ipexpert.com

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 www.PlatinumPlacement.com

___
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Re: [OSL | CCIE_Voice] AAR and uccx

2013-07-09 Thread jainpiyush2022
Hello Ram,

You can assign Hq device pool and location setting to cti route point and cti 
ports..
And assign site c device pool and location to site c phones... 

Regards,
Piyush Jain

Sent from my android device.



-Original Message-
From: Ramcharan Arya ramcharan.a...@gmail.com
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Sent: Tue, 09 Jul 2013 7:59 AM
Subject: [OSL | CCIE_Voice] AAR and uccx

Hi,
I have a CTI Route point 4000 and two CTI port 410x and 410y



SiteC Phone 1 and SC Phone 2 are in CSQ which is assign to application and
associated with trigger.

Due to  RSVP when call exceed ip rsvp bandwidth call  from uccx to site
Phones should use AAR and to over PSTN.

My doubts are .

What should be local and device pool of CTI ports so it should work in AAR
when PSTN caller make call to CTI route point number 4000.

Can someone please advice about this.

Thanks,
Ramcharan Arya
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Re: [OSL | CCIE_Voice] access to cli in lab

2013-07-08 Thread jainpiyush2022
Yes we have..

Thanks
Piyush Jain

Sent from my android device.



-Original Message-
From: Ajay Viswanath ajayviswan...@yahoo.co.in
To: ccie_voice@onlinestudylist.com
Sent: Mon, 08 Jul 2013 4:50 PM
Subject: [OSL | CCIE_Voice] access to cli in lab

Hi,

Do we have access to the server cli in the lab. Its vital for troubleshooting 
dbissues.

Thanks
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Re: [OSL | CCIE_Voice] Access list for cue traffic marking

2013-07-07 Thread jainpiyush2022
Steve, you absolutely make sense that traffic for cue can be marked on router 
(site c) on which cue module is connected when it goes out on wan link.. and 
then on the trunk port on hq switch we would have trust statement.

However the question in lab expect us to mark the cue traffic on hq switch on 
the port connected to sub cucm.. so the above solution won't work.. right?

Thanks and regards,
Piyush Jain

Sent from my android device.



-Original Message-
From: sbar...@mystictraveler.net
To: LorenzLGRC lorenzl...@gmail.com, Piyush Jain jainpiyush2...@ymail.com
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Sent: Mon, 08 Jul 2013 6:23 AM
Subject: RE: [OSL | CCIE_Voice] Access list for cue traffic marking

Maybe I am missing something so please forgive me, and to recap, the question 
was LAN QoS and CUE (not WAN).


The example below (which is pretty much out of the SRND)  will correctly mark 
the traffic, but only going out the serial port.   It seems to me that you 
would want to mark the traffic inbound from the CUE module to the router in 
which it resides  so that no matter how the traffic exits the router it will be 
handled correctly.  Can you mark the traffic as it leaves the AIM module and is 
passed to the router? 


As far as the policy map on the serial port, wouldn't we want to see all 
traffic correctly prioritized not just the CTI-QBE to answer the question 
correctly if we were to look at the WAN QoS?


I assume for traffic leaving on an LAN port to a switch, the switch would have 
the appropriate trust statements and since we marked on the packets as they 
transition from the AIM to the router prioritization and re-marking would not 
be an issue?


Steve

 Original Message 
Subject: Re: [OSL | CCIE_Voice] Access list for cue traffic marking
From: LorenzLGRC lorenzl...@gmail.com
Date: Sun, July 07, 2013 5:25 am
To: Piyush Jain jainpiyush2...@ymail.com
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com

Hello,

you can use something like this:

access-list 101 permit tcp host a.b.c.d any eq 2748 ! class-map match-all 
cti-qbe match access-group 101 ! policy-map cti-qbe class cti-qbe set dscp af31 
bandwidth 20 ! interface Serial0/1 service-policy output cti-qbe



On Sun, Jul 7, 2013 at 6:06 AM, Piyush Jain jainpiyush2...@ymail.com wrote:

Hi Guys,


I am trying to understand how we can mark CUE traffic on HQ Switch to implement 
LAN QOS.


I have come up with the below solution.


ip access-list extended name CUE

 permit tcp host 142.100.64.12 host 142.1.66.253 eq 2748



class-map match-any CUE-CLASS

 match access group name CUE


policy-map CUE-POLICY

 class CUE-CLASS

  set ip dhcp CS3


int fa 1/0/4

 description * CONNECTED TO SUB CUCM ***

 service policy input CUE-POLICY


In above config, 142.100.64.12 is SUB CUCM, 142.1.66.253 is CUE on SC router. 

Explanation: Since we are applying service policy in incoming direction on 
switch port connected to CUCM, so the source port number (of CUCM) can be 
anything but destination port number (i.e for CUE) should be 2748 (JTAPI port).


Any advice or inputs are most welcome.

 

Cheers !!
Piyush Jain



___
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___
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www.ipexpert.com

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www.PlatinumPlacement.com 

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Re: [OSL | CCIE_Voice] Cue call failed in srst mode

2013-07-07 Thread jainpiyush2022
Have you configured ccn subsystem and trigger for sip on cue? 

Thanks and regards,
Piyush Jain

Sent from my android device.



-Original Message-
From: Rrcrumm rrcr...@yahoo.com
To: Karen Johnson karen.johnson...@yahoo.ca
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com, sanity 
insanity networksanitytoinsan...@gmail.com
Sent: Mon, 08 Jul 2013 8:10 AM
Subject: Re: [OSL | CCIE_Voice] Cue call failed in srst mode

Dial peer and trigger

On Jul 7, 2013, at 4:22 PM, Karen Johnson karen.johnson...@yahoo.ca wrote:

 
 Yes
 
 From: Rrcrumm rrcr...@yahoo.com; 
 To: Karen Johnson karen.johnson...@yahoo.ca; 
 Cc: sanity insanity networksanitytoinsan...@gmail.com; 
 ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com;  

 Subject: Re: [OSL | CCIE_Voice] Cue call failed in srst mode 
 Sent: Sun, Jul 7, 2013 6:49:41 PM 
 
 Do you have the voicemail command under telephony-service?
 
 Hath
 Rc
 
 On Jul 7, 2013, at 11:04 AM, Karen Johnson karen.johnson...@yahoo.ca wrote:
 
 
 Hi folks,
 
 My cue is working fine in normal mode. If i switch to srst mode it failed 
 and busy tone. Dialpeer and codec g711 , sip-ua ip show correct.
 any idea what to tshoot and command to check?
 
 K
 
 From: sanity insanity networksanitytoinsan...@gmail.com; 
 To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com; 
 Subject: [OSL | CCIE_Voice] CME srst best practices.. 
 Sent: Sat, Jul 6, 2013 4:29:43 PM 
 
 hi Guys,
 
 In srst I use the following config...
 
 telephony-service
 srst mode auto-provision all
 srst dn line-mode octo
 
 
 1)Do I also need to configure srst dn template  srst ephone template ?
 
 2)what are best practices for setting up the cue in srst mode ? If possible 
 include details of..
 -mwi
 -Xfer to VM - button 6 of phone 1 is pressed
 
 -MJ
 ___
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 visit www.ipexpert.com
 
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 www.PlatinumPlacement.com
___
For more information regarding industry leading CCIE Lab training, please visit 
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Re: [OSL | CCIE_Voice] default device pool and css of already integrated devices!!

2013-07-06 Thread jainpiyush2022
Use default device pool for HQ phones and application servers like uccx, uc and 
presence.. this is the best way to do in lab.

Thanks and regards,
Piyush Jain

Sent from my android device.



-Original Message-
From: Drake J jdrake...@gmail.com
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Sent: Sun, 07 Jul 2013 6:19 AM
Subject: [OSL | CCIE_Voice] default device pool and css of already integrated 
devices!!

Hi Guys,

1) Could you clarify if we need to change the default device pool   CSS of
the devices already integrated such as unity connection , uccx , CUPS?

2)What would be the implications if we did make the above change?

3) If we were to retain the default device pool of the above devices then
do  we need to have all other sites setup to have a region relation with
the default region as g711ulaw?

-Drake
___
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[OSL | CCIE_Voice] Access list for cue traffic marking

2013-07-06 Thread jainpiyush2022
Hi guys,

I am trying to understand how we can mark cue traffic on ha switch..

I have come up with the below acl.

Ip access-list extended

Thanks and regards,
Piyush Jain

Sent from my android device.

___
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Re: [OSL | CCIE_Voice] directory handler in Unity connection

2013-07-03 Thread jainpiyush2022
I think you don't have first name defined on the end user page of cucm.. you 
might have defined only last name.check this first..

Thanks and regards,
Piyush Jain

Sent from my android device.



-Original Message-
From: Karen Johnson karen.johnson...@yahoo.ca
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Sent: Wed, 03 Jul 2013 11:16 AM
Subject: [OSL | CCIE_Voice] directory handler in Unity connection


folks,

when i use Directory handler  in CUC and press 'First Name. CUC can't find it. 
 is that anticipated and if not what i missed in config?
- what is the keys to switch to extension search ?


tks
K___
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Re: [OSL | CCIE_Voice] directory handler in Unity connection

2013-07-03 Thread jainpiyush2022
Delete from cuc and then re import.

Best Regards,
Piyush Jain

Sent from my android device.



-Original Message-
From: Karen Johnson karen.johnson...@yahoo.ca
To: jainpiyush2...@ymail.com jainpiyush2...@ymail.com, 
ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Sent: Wed, 03 Jul 2013 6:33 PM
Subject: Re: [OSL | CCIE_Voice] directory handler in Unity connection

yes, i have not assigned first name, now i am going to re-import from UCM user, 
but can't 

- do you know how to re-import UCM user to CUC ?

tks
K





 From: jainpiyush2...@ymail.com jainpiyush2...@ymail.com
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com; Karen 
Johnson karen.johnson...@yahoo.ca 
Sent: Tuesday, July 2, 2013 11:53:15 PM
Subject: Re: [OSL | CCIE_Voice] directory handler in Unity connection
 


I think you don't have first name defined on the end user page of cucm.. you 
might have defined only last name.check this first..

Thanks and regards,
Piyush Jain

Sent from my android device.



-Original Message-
From: Karen Johnson karen.johnson...@yahoo.ca
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Sent: Wed, 03 Jul 2013 11:16 AM
Subject: [OSL | CCIE_Voice] directory handler in Unity connection



folks,

when i use Directory handler  in CUC and press 'First Name. CUC can't find it. 
 is that anticipated and if not what i missed in config?
- what is the keys to switch to extension search ?


tks
K___
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Re: [OSL | CCIE_Voice] CUE/UCCX Integration

2013-05-21 Thread jainpiyush2022
This is what I follow For ccie lab,  i keep all the phones in none partition so 
that everything has access to the phones.
Also I keep uccx/cue route point and CTI port in none partition and assign none 
CSS to cue/ uccx route point and CTI port.

This works for me very well.

Thanks and regards,
Piyush Jain

Sent from my android device.



-Original Message-
From: ccie_voice-requ...@onlinestudylist.com
To: ccie_voice@onlinestudylist.com
Sent: Tue, 21 May 2013 9:37 PM
Subject: CCIE_Voice Digest, Vol 87, Issue 65

Send CCIE_Voice mailing list submissions to
ccie_voice@onlinestudylist.com

To subscribe or unsubscribe via the World Wide Web, visit
http://onlinestudylist.com/mailman/listinfo/ccie_voice
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ccie_voice-ow...@onlinestudylist.com

When replying, please edit your Subject line so it is more specific
than Re: Contents of CCIE_Voice digest...


Today's Topics:

   1. CUE/UCCX Integration (Nicolas MICHEL)


--

Message: 1
Date: Tue, 21 May 2013 14:23:31 +0200
From: Nicolas MICHEL mcl.nico...@gmail.com
To: OSL Voice ccie_voice@onlinestudylist.com, Nicolas MICHEL
mcl.nico...@gmail.com
Subject: [OSL | CCIE_Voice] CUE/UCCX Integration
Message-ID: 519b6743.2010...@gmail.com
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Hey Guys.

I had some trouble this weekend with my CUE integration with CUCM and I 
would like to know how to do it correctly while doing it fast ... (no 
best practices here I know :) )

This is how I see things.

-Create CTI RP in the none partition (accessible to anyone) with a CSS 
that can see CTI ports (let's say CSS_CTI_RP that includes PT_CTI_PORTS)

- Create 2 CTI Ports that are in the PT_CTI_PORTS with a CSS that can 
see directly phones (CSS_INTERNAL that include PT_INTERNAL)

-Create a user that has CTI standard enabled and controlled devices : 
CTI-RP and CTI_Ports

-Create a voicemail pilot that is equal to the CTI RP in the none partition

-Create a Voicemail profile that include the newly created voicemail 
pilot for BR2

-Apply Voicemail profile to the phones *reset them*

Then on the CUE.

-Check the license and upload new license if not OK

-Go Offline and restore factory default

- Wizard install: Configure CUCM IP/CTI user/ etc etc and imports users .


Check if the CTI ports are registered on the CUCM
Check if the CTI RP is registered on the CUCM

Check by pressing the voicemail button on the phone...



My main concern is about the CSS/PT configuration ...

Can someone please enlighten me ? That would be awesome !

Thanks

Nic


--

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End of CCIE_Voice Digest, Vol 87, Issue 65
**
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Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 87, Issue 38

2013-05-16 Thread jainpiyush2022
Hello Ikizoo,

Your service name under application is cmm and under dial peer you are invoking 
ccm so incoming dial peer does not find the service and call fails. 
Also in the debugs you can see this error..


RELEASE_COMP pd = 8  callref = 0x80A1 Cause i = 0x80BF - Service/option 
not available, unspecified


Thanks and regards,
Piyush Jain

Sent from my android device.



-Original Message-
From: ccie_voice-requ...@onlinestudylist.com
To: ccie_voice@onlinestudylist.com
Sent: Thu, 16 May 2013 11:01 AM
Subject: CCIE_Voice Digest, Vol 87, Issue 38

Send CCIE_Voice mailing list submissions to
ccie_voice@onlinestudylist.com

To subscribe or unsubscribe via the World Wide Web, visit
http://onlinestudylist.com/mailman/listinfo/ccie_voice
or, via email, send a message with subject or body 'help' to
ccie_voice-requ...@onlinestudylist.com

You can reach the person managing the list at
ccie_voice-ow...@onlinestudylist.com

When replying, please edit your Subject line so it is more specific
than Re: Contents of CCIE_Voice digest...


Today's Topics:

   1. Cucm log (Dharambir kumar varma)
   2. Cucm log (Dharambir kumar varma)
   3. Re: MVA Fast Busy (ikizoo hello)
   4. Re: MVA Fast Busy (Abdullin Kamil)
   5. Re: MVA Fast Busy (GRASSMUGG Christoph)


--

Message: 1
Date: Thu, 16 May 2013 10:00:55 +0530
From: Dharambir kumar varma dharambi...@gmail.com
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Cucm log
Message-ID:
ca+iwkjsxty0ddkospu06mw4xqsuknqebbkos0nwguv-xqbx...@mail.gmail.com
Content-Type: text/plain; charset=iso-8859-1

Hi
Can we delete the audit logs on cucm which is seen on Rtmt tool.
Is it possible ...
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Message: 2
Date: Thu, 16 May 2013 10:05:26 +0530
From: Dharambir kumar varma dharambi...@gmail.com
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Cucm log
Message-ID:
CA+iWkJSBDGJQQk6E2TXjCXJptj4zWa3_ZgVy2Oc2XJ0vU=_...@mail.gmail.com
Content-Type: text/plain; charset=iso-8859-1

Hi
Can we delete audit logs on cucm
Which can be seen on Rtmt tool.
Is it possible
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Message: 3
Date: Wed, 15 May 2013 21:48:20 -0700
From: ikizoo hello ikiz...@hotmail.com
To: Josh Petro josh.pe...@gmail.com,
ccie_voice@onlinestudylist.comccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] MVA Fast Busy
Message-ID: bay172-w29a3a6888bf62b11239673ea...@phx.gbl
Content-Type: text/plain; charset=iso-8859-1

application service cmm http://10.10.210.10:8080/ccmivr/pages/IVRMainpage.vxml
dial-peer voice 2 pots service ccm
ccm --- cmm ?
Date: Wed, 15 May 2013 23:48:40 -0400
From: josh.pe...@gmail.com
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] MVA Fast Busy

Hi All,
I'm on Volume 1 section 5C which is asking for Mobile Vocie Access 
configuration. What I'm getting when I dial from the PSTN to the HQ Phone 2 is 
a fast busy after a few seconds of silence. 

I know this has been talked about in the past on this forum, but for the life 
of me I can't find it.
I've followed what was in the workbook exactly and I'm not getting anywhere. 
Now, I'm doing this on my home lab with IOS 12.4.22T5 and nothing else in the 
flash directory. Is there something that needs to be 'installed' on the router 
to allow it to see the vxml code on the CUCM? I've never been asked to do 
Mobile Voice Access before, so this is a first.

The CUCM is configured with the Mobile Voice Access number of 5999 and the 
gateway is configured for 4 digits inbound (plus I tested other inbound calls 
and they still work fine).

The Services parameters are correct and MVA is enabled. The voice-class codec 
has G711u and G729r8 and that's it. I'm thinking codec, but I'm not sure why.
I can't figure out what I missed.

Many thanks!Josh
The config and debug isdn q931 output is below if it helps. I didn't get the 
chance to get the voip ccapi debugs, so please let me know if you need them and 
I'll grab them tomorrow. 


!application service cmm http://10.10.210.10:8080/ccmivr/pages/IVRMainpage.vxml 
!!
dial-peer voice 5010 voip description MVA DID destination-pattern 5999 
voice-class codec 1 session target ipv4:10.10.210.10 incoming called-number . 
dtmf-relay h245-alphanumeric
 no vad!dial-peer voice 2 pots service ccm incoming called-number 2123945999 no 
digit-strip!
!HQ-RTR(config-dial-peer)#May 16 03:12:37.874: ISDN Se0/0/0:23 Q931: RX - 
SETUP pd = 8  callref = 0x00A1 Bearer Capability i = 0x8090A2   
  Standard = 

Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 87, Issue 30

2013-05-14 Thread jainpiyush2022
For early media, Mtp codec depends on your requirement. If your inter region 
codec requirement is g729 then define that in the mtp codec configuration and 
you don't need to use pass through..

 I have tested this in my lab..

Thanks and regards,
Piyush Jain

Sent from my android device.



-Original Message-
From: ccie_voice-requ...@onlinestudylist.com
To: ccie_voice@onlinestudylist.com
Sent: Mon, 13 May 2013 9:31 PM
Subject: CCIE_Voice Digest, Vol 87, Issue 30

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Today's Topics:

   1. Re: MRG for MTP, SIP Early Offer (FAISAL AL-EMAD)
   2. Cisco ip phone over internet (Dharambir kumar varma)
   3. Registering phone (Dharambir kumar varma)
   4. Re: h323 fast start (Kirill Groshev)


--

Message: 1
Date: Mon, 13 May 2013 09:14:13 +0300
From: FAISAL AL-EMAD eng_ale...@hotmail.com
To: Bill Lake whl...@gmail.com, Ben John benjoh...@hotmail.com
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] MRG for MTP, SIP Early Offer
Message-ID: dub123-w1493846c19fce1225b3a33fc...@phx.gbl
Content-Type: text/plain; charset=windows-1256

Dear Bill,

You are right, with Early offer we need MTP but in the configuration of MTP 
should enable codec g711ulaw and pass-through only?


Best Regards
Eng. Faisal alemadNetworks  UC Engineer

 Three things in life that make you a great person
1. Hard Work 2.  Sincerity  3.  success

Date: Sun, 12 May 2013 19:40:32 -0500
From: whl...@gmail.com
To: benjoh...@hotmail.com
CC: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] MRG for MTP, SIP Early Offer

Early offer requires media resources


Delayed offer does not require it


SIP Early Offer Support over Unified CM SIP Trunks


SIP negotiates media exchange by means of the Session Description 
Protocol (SDP), where one side offers a set of capabilities to which the
 other side answers, thus converging on a set of media characteristics. 
SIP allows the initial offer to be sent either by the caller in the 
initial INVITE message (Early Offer) or, if the caller chooses not to, 
the called party can send the initial offer in the first reliable 
response (Delayed Offer).


By default, Unified CM SIP trunks send the INVITE without an initial 
offer (Delayed Offer). In general SIP Delayed Offer is preferred for 
Unified CM SIP trunks because MTPs are not needed to establish a Delayed
 Offer call for voice, video, or encrypted media. If SIP Early Offer is 
desired, Unified CM has two configurable options to enable a SIP trunk 
to send the offer in the INVITE:


?Media Termination Point Required


?Early Offer Support for Voice and Video Calls (Insert MTP If Needed)


 


Media Termination Point Required


Enabling the Media Termination Point Required option on the SIP trunk assigns 
an MTP from the trunk's media resources group (MRG) to every outbound call. 
(See Figure 14-3.) This statically assigned MTP supports only the G.711 or 
G.729 codecs, thus limiting media to voice calls only.


Figure 14-3 SIP Early Offer with Media Termination Point Required





http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/trunks.html#wp1123289





On Sun, May 12, 2013 at 9:24 AM, Ben John benjoh...@hotmail.com wrote:




Hello All,
when configuring SIP Delay Offer do we have to configure MRG for MTP ? In one 
of IPExpert lab work book i have seen them configuring MRG for Early Offer. 
Please advise

 
Ben
  

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Message: 2
Date: Mon, 13 May 2013 13:54:21 +0530
From: Dharambir kumar varma dharambi...@gmail.com
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Cisco ip phone over internet
Message-ID: