Hi All,

In lab 8 we have to do FS with SiteB GW. If we use G711u codec for FS then it 
fails the call for HQ phones when they use SiteB GW for routing as backup route 
because HQ and SiteB uses G729 as inter region codec.

I tried to change the g711u to g729r8 in mtp configuration for FS and under 
outbound FS enable drop down option. Then in that case it is working for all 
the calls.

Any other solution anybody can think of for this FS issue..??

Regards,
Piyush Jain




________________________________
 From: "ccie_voice-requ...@onlinestudylist.com" 
<ccie_voice-requ...@onlinestudylist.com>
To: ccie_voice@onlinestudylist.com 
Sent: Tuesday, May 14, 2013 8:20:08 PM
Subject: CCIE_Voice Digest, Vol 87, Issue 32
 

Send CCIE_Voice mailing list submissions to
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Today's Topics:

   1. Re: CCIE_Voice Digest, Vol 87, Issue 30 (jainpiyush2...@ymail.com)
   2. Calling Name for MVA (CCIEing)
   3. Re: Calling Name for MVA (heshamcentr...@gmail.com)


----------------------------------------------------------------------

Message: 1
Date: Tue, 14 May 2013 11:17:46 +0530
From: jainpiyush2...@ymail.com
To: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 87, Issue 30
Message-ID: <b691de99-251f-441c-9818-54f948ea0d62.maildroid@localhost>
Content-Type: text/plain; charset="utf-8"

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

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. 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
                          

_______________________________________________

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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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:
    <CA+iWkJTR7wb6HRLiz9b_1jav2Ls9P6=Qz-sxGn+VVHExbs4=f...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Hi
Please help how to register ip phone over internet..
What parameters are required on cucm and phone.
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Message: 3
Date: Mon, 13 May 2013 18:20:44 +0530
From: Dharambir kumar varma <dharambi...@gmail.com>
To: "ccie_voice@onlinestudylist.com" <ccie_voice@onlinestudylist.com>
Subject: [OSL | CCIE_Voice] Registering phone
Message-ID:
    <CA+iWkJRghNbZgp2h+=CU2fzAv-5ZrieJWd3=0v0dnxhy2ba...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Hi
we r using cucm at hq.we have to register one 7941 phone connected with
broadband internet.please help the procedure to register.
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Message: 4
Date: Mon, 13 May 2013 17:44:30 +0200
From: Kirill Groshev <kgros...@gmail.com>
To: Hesham Abdelkereem <heshamcentr...@gmail.com>
Cc: "ccie_voice@onlinestudylist.com" <ccie_voice@onlinestudylist.com>
Subject: Re: [OSL | CCIE_Voice] h323 fast start
Message-ID:
    <calyaauibliwoi3tnc7kgnkmztj1-iys54m+wqm+kibntc3u...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Hi gentlemen,
I was getting the CallManager service traces and looking for incoming
andh225 setup messages with h245 elements. You will see the keyword
"faststart" in the trace as well as  offered codec. However, this approach
is quite time consuming.

Cheers,
Kirill

On Wednesday, May 8, 2013, Hesham Abdelkereem wrote:

> WOOOOOOOOOOOOOOOOOOOOW
> I just got that link earlier this morning as well as Suresh really helped.
> Bill , Thank you so much you've been such a great helpful man.
> I appreciate all your great efforts.
>
> Many Thanks,
> Hesham
>
> On 7 May 2013 15:35, Bill Lake <whl...@gmail.com <javascript:_e({},
> 'cvml', 'whl...@gmail.com');>> wrote:
>
>>  try this
>>
>> http://ciscovoip-amitr.blogspot.com/2011/04/fast-start-vs-slow-start.html
>>
>>
>>  On Tue, May 7, 2013 at 1:44 PM, Barrera, Hugo 
>><hugo.barr...@nexusis.com<javascript:_e({}, 'cvml', 
>>'hugo.barr...@nexusis.com');>
>> > wrote:
>>
>>>   Hi,****
>>>
>>> ** **
>>>
>>> Anyone guide me in the right direction how to read the debugs for h323
>>> fast start? ****
>>>
>>> ** **
>>>
>>> *thanks*****
>>>
>>> _______________________________________________
>>> 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 <http://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 <http://www.platinumplacement.com/>
>>
>
>
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End of CCIE_Voice Digest, Vol 87, Issue 30
******************************************
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Message: 2
Date: Tue, 14 May 2013 17:18:55 +0300
From: CCIEing <aboaz...@gmail.com>
To: "ccie_voice@onlinestudylist.com" <ccie_voice@onlinestudylist.com>
Subject: [OSL | CCIE_Voice] Calling Name for MVA
Message-ID:
    <CAEHpKkL1zRugxCjRS40-zLZWrmS=0dh4faiedsrgfjrppms...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Hi all,

I had the following problem , when I make a call from remote Destination to
the MVA IVR number, then to internal extension, the calling name for the
configured in the line for the remote destination profile does not appear
on the internal phone when it answered the call, this happen only when I
use the enable inbound faststart with my h323 gateway (where the MVA
configured) .

If I unchecked the inbound faststart the calling name appear.


any idea ?
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------------------------------

Message: 3
Date: Tue, 14 May 2013 14:50:02 +0000
From: heshamcentr...@gmail.com
To: "CCIEing" <aboaz...@gmail.com>,
    ccie_voice-boun...@onlinestudylist.com,
    "ccie_voice@onlinestudylist.com" <ccie_voice@onlinestudylist.com>
Subject: Re: [OSL | CCIE_Voice] Calling Name for MVA
Message-ID:
    
<377326133-1368543004-cardhu_decombobulator_blackberry.rim.net-477083750-@b5.c10.bise6.blackberry>
    
Content-Type: text/plain

Yes when you create your remote destination profile and associate it with the 
phone at the end of the page you will find a purple section that has alerting 
name ascii and external number mask u have to rewrite that again and its not 
configured by default from associating the same line with the phone 
Hope this helps

Thanks
Sent via BlackBerry from T-Mobile

-----Original Message-----
From: CCIEing <aboaz...@gmail.com>
Sender: ccie_voice-boun...@onlinestudylist.com
Date: Tue, 14 May 2013 17:18:55 
To: ccie_voice@onlinestudylist.com<ccie_voice@onlinestudylist.com>
Subject: [OSL | CCIE_Voice] Calling Name for MVA

_______________________________________________
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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http://onlinestudylist.com/mailman/listinfo/ccie_voice


End of CCIE_Voice Digest, Vol 87, Issue 32
******************************************
_______________________________________________
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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