Re: [OSL | CCIE_Voice] RSVP Problem

[Vasanth]

Yes. In that case you can go with the approach you have taken.

I had tough time when doing it in a different lab scenario and then finally
removing the QoS policy fixed my problem and then did reverse engineering
of the QoS policy(with help of CCM Traces) to find out that fragmentation
is causing the problem. If the ask is to configure LFI and RSVP on the same
link then you might be using a different version of IOS/CCM.

As WAN QoS question I would alway start with auto qos and work through the
class-map,map-class,policy-map to meet the requirement.

Cheers,
Vasanth


Regards,
Vasanth


On Tue, Jun 11, 2013 at 2:40 AM, Martin Sloan wrote:

> Hi Vasanth,
>
> Thanks for the reply.  I was able to get this working by
> removing/re-adding the RSVP MTP association to the MRG in CUCM.  Calls are
> working fine with RSVP now.  About the fragmentation, it's required as part
> of the next task for WAN QoS with LFI between HQ->BR1 so I don't think I
> can avoid that part.  Do you agree?
>
> I posted a similar question to the group in regard to setting up the LFI
> for these tasks.  I've been using auto qos because it creates all of the
> class-maps and calculates the fragment size for me, so there's no digging
> for the information within Cisco docs.  If you're interested, check out my
> email from 6/1 titled 'Advice or opinions on Vol 2 Lab 4 Task 5.1' and let
> me know what you think.
>
> Thanks,
> Marty
>
>
> On Mon, Jun 10, 2013 at 3:46 PM, Vasanth  wrote:
>
>>
>> On Mon, Jun 10, 2013 at 9:30 PM, 
>> wrote:
>>
>>> command under the dspfarm profile as well, it didn't copy over in my
>>> email:
>>>
>>
>> Hi Martin,
>>
>> You have auto qos enabled for 384 bandwidth frame-relay link.
>>
>> This would bring in frame-relay fragmentation of packets.
>>
>> auto qos would configure frame-relay fragment size of 480 bytes.
>>
>> This causes the ccm not able to parse the rsvp response from the Router
>> (MTP) to call manager.
>>
>> If you can remove the frame-relay fragment command and check RSVP it
>> should work.
>>
>> Regards,
>> Vasanth
>>
>> ___
>> 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] CUE & NTP - Best practices

[Michael.Sears]

Hey MJ,

DBreplication issues are very time consuming to fix and troubleshoot and out of 
my four attempts I didn't have any issues with replication.  I know several 
others who are on their 7th and 8th attempts and they too have NOT had 
DBreplication issues, especially with regards to NTP.  There are two ways that 
I'm aware of to check the status, Reports>Database Summary and RTMT, although 
sometimes they are incorrect.

I did ask the proctor in RTP if there was a DBreplication issue was that 
considered part of the lab.  He responded that you are expected to fix it 
although I haven't experienced it.

A good question to the group I think would be just to ask if anyone has had 
DBreplicaation issues while sitting the lab.  Also, maybe there is a 
DBreplication guru out there that can speak to DBreplication issues better than 
myself.  In my lab with severe DBreplication issues it was usually quicker to 
revert to a good snapshot then spend much time on trying to fix it.

There is no need to restart any services on the Publisher or Subscriber when 
you add NTP to the Publisher.  The Subscriber will get its time from the 
Publisher and the Publisher will get its time from HQ1 loopback if that's what 
the question calls for.  No need to reset anything.  In the case of the HQ1 
loopback it gets it time from the PSTN.  In my lab I used  the atomic clocks in 
Boulder, CO and my configuration on my PSTN router for NTP looked like this:

PSTN#
ntp source Loopback1
ntp master 1
ntp update-calendar
ntp server 132.163.4.101 prefer burst iburst
ntp server 132.163.4.102 burst iburst
ntp server 132.163.4.103 burst iburst

This assumes that you have internet access from the PSTN router.  To trouble 
shoot ntp I use the following:

PSTN-WAN#debug ntp ?
  adjustNTP clock adjustments
  all  NTP all debugging on
  core   NTP core messages
  eventsNTP events
  packetNTP packet debugging
  refclock NTP refclock messages

PSTN#debug ntp all

You can also use "utils diagnose test" to check for any dns issues or ntp 
issues related to DBreplication.

admin:utils diagnose test

test - ntp_reachability : Passed
test - ntp_clock_drift  : Passed
test - ntp_stratum: Passed

If you do have a problem in the LAB with DBreplication better to fix it 
sometimes DB repair works than try to determine the root cause.  Otherwise you 
won't finish your lab.  I didn't focus much on DBreplication when I took my 
attempts, maybe I should have, but I passed without any issues with NTP or 
DBreplication issues.

Well not sure if any of this helps, but good luck in your studies.  You may 
want to NOT focus so much on this one topic although your questions are valid 
and the information is good to know.

--ms

Michael Sears, CCIE(V)#38404
Cisco Certified Unfied Communications Computing Systems Specialist
E911 Insrastructure Specialist
[3-2-2013 3-02-38 PM]   [UCS SPECIALIST1]
"Designing and Implementing Cisco Unified Communications on Unified Computing 
Systems"

From: sanity insanity [mailto:networksanitytoinsan...@gmail.com]
Sent: Monday, June 10, 2013 6:38 PM
To: Sears, Michael (msears)
Cc: ccie_voice@onlinestudylist.com
Subject: Re: CUE & NTP - Best practices

hi Guys,
Still waiting to hear back

Thanks again

On Sun, Jun 9, 2013 at 10:22 PM, sanity insanity 
mailto:networksanitytoinsan...@gmail.com>> 
wrote:
hi Guys,
Thanks for your replies.
If the DB replication does go out of sync is there any troubleshooting step 
that can be executed  .  Hope it does not take too much  time to sync since we 
would then loose the time to complete  other tasks in the process.
1) Also does restart of NTP on Publisher and subscriber fix all issues related 
to DB replication caused due to NTP?

2) Is it recommended that we restart NTP service on Publisher and subscriber 
after adding NTP server to the Publisher?

3) What can be done to determine the cause of the issue?
-MJ


On Wed, Jun 5, 2013 at 10:55 PM, Sears, Michael (msears) 
mailto:michael.se...@compucom.com>> wrote:
> 1) which is the best way to bind the service module to the router is
> it required to bind it to the loopback of the router or  the interface
> voice VLAN.
>
>Answer:  It depends on the question it could be either the Loopback or the
>Voice VLAN it should say in the LAB.  In my practice I use both alternating,
>one lab I'll use the Voice VLAN and then on another run through the lab
>I'll use the Loopback.
>
>Note that people have had issues using the Loopback so best to figure it out
>before setting the LAB.
>
> 2) On my HQ router I am configuring  it to sync with a back NTP
> server. I am also required to sync the CUCM publisher with the
> loopback of this HQ router . Here are my questions...
>
> a) I have the following configuration on the HQ router...
> ntp source Loopback0
> ntp server 177.26.1.100
>
>HQ ntp configuration:
>ntp source Loopback0
>ntp server 177.26.

Re: [OSL | CCIE_Voice] CUE & NTP - Best practices

[sanity insanity]

hi Guys,

Still waiting to hear back


Thanks again


On Sun, Jun 9, 2013 at 10:22 PM, sanity insanity <
networksanitytoinsan...@gmail.com> wrote:

> hi Guys,
>
> Thanks for your replies.
>
> If the DB replication does go out of sync is there any troubleshooting
> step that can be executed  .  Hope it does not take too much  time to sync
> since we would then loose the time to complete  other tasks in the process.
>
> 1) Also does restart of NTP on Publisher and subscriber fix all issues
> related to DB replication caused due to NTP?
>
> 2) Is it recommended that we restart NTP service on Publisher and
> subscriber after adding NTP server to the Publisher?
>
> 3) What can be done to determine the cause of the issue?
>
> -MJ
>
>
>
>
> On Wed, Jun 5, 2013 at 10:55 PM, Sears, Michael (msears) <
> michael.se...@compucom.com> wrote:
>
>> > 1) which is the best way to bind the service module to the router is
>> > it required to bind it to the loopback of the router or  the interface
>> > voice VLAN.
>> >
>> >Answer:  It depends on the question it could be either the Loopback or
>> the
>> >Voice VLAN it should say in the LAB.  In my practice I use both
>> alternating,
>> >one lab I'll use the Voice VLAN and then on another run through the lab
>> >I'll use the Loopback.
>> >
>> >Note that people have had issues using the Loopback so best to figure it
>> out
>> >before setting the LAB.
>> >
>> > 2) On my HQ router I am configuring  it to sync with a back NTP
>> > server. I am also required to sync the CUCM publisher with the
>> > loopback of this HQ router . Here are my questions...
>> >
>> > a) I have the following configuration on the HQ router...
>> > ntp source Loopback0
>> > ntp server 177.26.1.100
>> >
>> >HQ ntp configuration:
>> >ntp source Loopback0
>> >ntp server 177.26.1.100 burst iburst
>> >
>> > 3)  On CUCM PUB  I have added the NTP server  and given it the ip
>> > address of loopback of HQ router. Now my questions are...
>> >
>> > 1) Is a reboot of CUCM required?  No
>> > 2) If the ntp does not sync will my DB replication break after a few
>> hours?
>> >
>> >Answer:  If the NTP Server does not synchronize need to determine why.
>>  When
>> >I've had issues in the past with NTP not synchronizing it did not break
>> >dbreplication over a period of hours.  Although I suppose there's that
>> possibility.
>> >
>> >
>> >Michael Sears, CCIE(V)#38404
>>
>>
>>
>
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[OSL | CCIE_Voice] 3750 IOS version in lab

[Brian Valentine]

Unless it breaks NDA, can someone please tell me what version of IOS is
running on the 3750 switch in the lab exam?  The blueprint is generic in
that it just says 12.2 Mainline.

Thanks,

Brian Valentine
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Re: [OSL | CCIE_Voice] RSVP Problem

[Bill Lake]

Hi Marty,

That is a great lesson for you, the more you practice this stuff, the more
you learn what can go wrong, how to ID it and fix it fast.

Bill


On Mon, Jun 10, 2013 at 12:01 PM, Martin Sloan wrote:

> Suresh,
>
> Thank you for the help!  I did have the 'ip rsvp bandwidth' setting under
> the physical interfaces as well.  As a last-ditch effort I removed the
> HQ-RSVP MTP from the MRG, inserted a generic MTP and saved, then reverted
> that change and put the HQ-RSVP MTP back in and saved.voila!  It now
> works.  I don't know what that accomplished which a system reboot did not,
> but it's a lesson learned in troubleshooting.  Before I hack through system
> traces, I should try some quicker fixes first.  It would have been faster
> had I just ripped it all out and rebuilt it.  It's hard to not get bogged
> down sometimes when you're in the weeds.
>
> Thanks again for the help.
> Marty
>
>
> On Mon, Jun 10, 2013 at 12:15 PM, Suresh Bhandari wrote:
>
>> What happens when you make a call from HQ to BR1 phones? If phones ring,
>> check if show sccp connections has expected output - that is, there is a
>> reservation of 40K bandwidth for each ringing phones.
>>
>> You have partial configuration included here. Can you make sure that you
>> have configured physical interfaces, Serial0/1/0 (from your config), on
>> both routers to have ip rsvp bandwidth?
>>
>>
>>
>> On Mon, Jun 10, 2013 at 8:24 PM, Martin Sloan wrote:
>>
>>> Just re-read my configs, please note that BR1 *does* have the 'rsvp'
>>> command under the dspfarm profile as well, it didn't copy over in my email:
>>>
>>>
>>> dspfarm profile 1 mtp
>>>  codec pass-through
>>>  codec g729r8
>>>  rsvp
>>>  maximum sessions software 4
>>>  associate application SCCP
>>>
>>>
>>> On Mon, Jun 10, 2013 at 10:28 AM, Martin Sloan 
>>> wrote:
>>>
 I'm running into an issue getting RSVP to work on Vol 2 Lab 5 and would
 like some advice on what I might have missed.

 The requirement is to allow an equal number of calls (2) over redundant
 links from HQ->BR1 and 4 calls from HQ->BR2.  Also, RSVP should use 'video
 desired' allowing calls to proceed as audio only when there is not enough
 bandwidth for audio and video.  Just using HQ->BR1 as an example, so far I
 have configured:

 - bandwidth statements on Serial sub-interfaces:

 HQ:

 interface Serial0/1/0.1 point-to-point
  bandwidth 384
  ip address 10.10.111.1 255.255.255.252
  ip ospf mtu-ignore
  snmp trap link-status
  frame-relay interface-dlci 201
   class AutoQoS-FR-Se0/1/0-201
   auto qos voip
  ip rsvp bandwidth 64
 !
 interface Serial0/1/0.2 point-to-point
  bandwidth 384
  ip address 10.10.111.5 255.255.255.252
  ip ospf mtu-ignore
  snmp trap link-status
  frame-relay interface-dlci 211
   class AutoQoS-FR-Se0/1/0-201
  ip rsvp bandwidth 64

 BR1:

 interface Serial0/1/0.1 point-to-point
  bandwidth 384
  ip address 10.10.111.2 255.255.255.252
  ip ospf mtu-ignore
  snmp trap link-status
  frame-relay interface-dlci 101
   class AutoQoS-FR-Se0/1/0-201
  ip rsvp bandwidth 64
 !
 interface Serial0/1/0.2 point-to-point
  bandwidth 384
  ip address 10.10.111.6 255.255.255.252
  ip ospf mtu-ignore
  snmp trap link-status
  frame-relay interface-dlci 111
   class AutoQoS-FR-Se0/1/0-201
  ip rsvp bandwidth 64

 - Software MTP on the router:

 HQ:

 sccp local Loopback0
 sccp ccm 10.10.210.10 identifier 2 priority 2 version 7.0
 sccp ccm 10.10.210.11 identifier 1 priority 1 version 7.0
 sccp
 !
 sccp ccm group 1
  bind interface Loopback0
  associate ccm 1 priority 1
  associate ccm 2 priority 2
  associate profile 1 register HQ-RSVP
 !
 dspfarm profile 1 mtp
  codec pass-through
  codec g729r8
  rsvp
  maximum sessions software 8
  associate application SCCP

 BR1:

 sccp local Loopback0
 sccp ccm 10.10.210.11 identifier 1 priority 1 version 7.0
 sccp ccm 10.10.210.10 identifier 2 priority 2 version 7.0
 sccp
 !
 sccp ccm group 1
  bind interface Loopback0
  associate ccm 1 priority 1
  associate ccm 2 priority 2
  associate profile 1 register BR1-RSVP
 !
 dspfarm profile 1 mtp
  codec pass-through
  codec g729r8
  maximum sessions software 4
  associate application SCCP

 - Configured both MTP's in CUCM (both are registered)
 - Placed MTP's into their own MRG
 - Placed the RSVP MRG at the bottom of the MRGL for each site
 - Set Locations RSVP settings to mandatory (video desired)
 - Inter-region settings for HQ/BR1 is set to g729
 - HQ Device Pool contains HQ Location and Region
 - BR1 Device Pool contains BR1 Location and Region
 - HQ Device Pool is assigned to the phone placing the call to BR1
 - 

Re: [OSL | CCIE_Voice] RSVP Problem

[Martin Sloan]

Hi Vasanth,

Thanks for the reply.  I was able to get this working by removing/re-adding
the RSVP MTP association to the MRG in CUCM.  Calls are working fine with
RSVP now.  About the fragmentation, it's required as part of the next task
for WAN QoS with LFI between HQ->BR1 so I don't think I can avoid that
part.  Do you agree?

I posted a similar question to the group in regard to setting up the LFI
for these tasks.  I've been using auto qos because it creates all of the
class-maps and calculates the fragment size for me, so there's no digging
for the information within Cisco docs.  If you're interested, check out my
email from 6/1 titled 'Advice or opinions on Vol 2 Lab 4 Task 5.1' and let
me know what you think.

Thanks,
Marty


On Mon, Jun 10, 2013 at 3:46 PM, Vasanth  wrote:

>
> On Mon, Jun 10, 2013 at 9:30 PM, 
> wrote:
>
>> command under the dspfarm profile as well, it didn't copy over in my
>> email:
>>
>
> Hi Martin,
>
> You have auto qos enabled for 384 bandwidth frame-relay link.
>
> This would bring in frame-relay fragmentation of packets.
>
> auto qos would configure frame-relay fragment size of 480 bytes.
>
> This causes the ccm not able to parse the rsvp response from the Router
> (MTP) to call manager.
>
> If you can remove the frame-relay fragment command and check RSVP it
> should work.
>
> Regards,
> Vasanth
>
> ___
> 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] RSVP Problem

[Vasanth]

On Mon, Jun 10, 2013 at 9:30 PM, wrote:

> command under the dspfarm profile as well, it didn't copy over in my email:
>

Hi Martin,

You have auto qos enabled for 384 bandwidth frame-relay link.

This would bring in frame-relay fragmentation of packets.

auto qos would configure frame-relay fragment size of 480 bytes.

This causes the ccm not able to parse the rsvp response from the Router
(MTP) to call manager.

If you can remove the frame-relay fragment command and check RSVP it should
work.

Regards,
Vasanth
___
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Re: [OSL | CCIE_Voice] LAB GRADED WITH ALL 0%.

[Michael.Sears]

Ravi,

All 0% does sound a little strange.  I've never heard of anyone getting all 0% 
before.  Possibly your configurations were wiped out somehow.  If you feel that 
your lab was not graded correctly or there was a problem with grading call the 
following number and open a case with the CCIE Certifications Team.  
1.800.553.6387 #4 #1
--ms
Michael Sears, CCIE(V)#38404
Cisco Certified Unfied Communications Computing Systems Specialist
E911 Infrastructure Specialist
>
>
> Dear All,
>
> I got CCIE lab exam for the first time and I am really unsatisfied 
> about the marks I got. unfortunately I was not able to complete my lab 
> properly but I have configured few sections in good manner. 
> Unfortunately I didn't knew that I removed a cable from a phone and 
> replugged due to phones were not registered with CUCM correctly. when 
> I got a result sheet it was really strange and they given me 0% for 
> each and every section. So is there any way to appeal the exam to 
> cisco and get another chance to attain this valuable certification. I 
> know that I am technically competent. but now I am totally depressed.
>
> Thanks,
> Ravi,

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Re: [OSL | CCIE_Voice] Sip Troubleshooting

[ikizoo hello]

1) The first line "//SIP/SIPTcp...Outgoing SIP TCP"  >>> this is not part 
of SIP message[ccm message]2) ""  
>>> this indicate tcp as transport protocol  For Early Offer 
1)Content-Type: application/sdp >> this indicate message include SDP which 
means EO2)v=0
-ikizoo
Date: Mon, 10 Jun 2013 17:53:46 +0545
From: bring...@gmail.com
To: ccie2...@gmail.com
CC: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Sip Troubleshooting

For me its the Content-Length:xxx with non-zero value.


On Fri, Jun 7, 2013 at 5:13 PM, ccie2k12  wrote:














I know the below message is clearly telling us that calling
side is doing SIP Early offer and using TCP as the transport protocol

but from the below message where do you think we should mark
 for TCP and Early Offer

 

TCP 

 

1) The first line "//SIP/SIPTcp...Outgoing SIP
TCP"

2) ""

 

For Early Offer

 

1)Content-Type: application/sdp

2)v=0

 

CCM|//SIP/SIPTcp/wait_SdlSPISignal:
Outgoing SIP TCP message to 151.1.2.2 on port 5060 index 6 

INVITE sip:911@151.1.2.2:5060 SIP/2.0

Date: Fri, 07 Jun 2013 04:09:56 GMT

Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK,
UPDATE, REFER, SUBSCRIBE, NOTIFY

From: "HQPH1" 
;tag=06bb5c4b-f416-48f6-9225-822962b6cef1-29652495

Allow-Events: presence, kpml

P-Asserted-Identity: "HQPH1" 

Supported: timer,resource-priority,replaces

Min-SE:  1800

Remote-Party-ID: "HQPH1" 
;party=calling;screen=yes;privacy=off

Content-Length: 210

User-Agent: Cisco-CUCM7.0

To: 

Contact:


Expires: 180

Content-Type:
application/sdp

Call-ID: 1b618780-1b115d14-5-a0a01b1@151.1.10.10

Via: SIP/2.0/TCP 151.1.10.10:5060;branch=z9hG4bK77edd7cbb

CSeq: 101 INVITE

Session-Expires:  1800

Max-Forwards: 70

 

v=0

o=CiscoSystemsCCM-SIP 2000 1 IN IP4 151.1.10.10

s=SIP Call

c=IN IP4 151.1.254.1

t=0 0

m=audio 16416 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=ptime:20

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15







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-- 
Suresh Bhandari



___
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www.PlatinumPlacement.com ___
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Re: [OSL | CCIE_Voice] RSVP Problem

[Martin Sloan]

Suresh,

Thank you for the help!  I did have the 'ip rsvp bandwidth' setting under
the physical interfaces as well.  As a last-ditch effort I removed the
HQ-RSVP MTP from the MRG, inserted a generic MTP and saved, then reverted
that change and put the HQ-RSVP MTP back in and saved.voila!  It now
works.  I don't know what that accomplished which a system reboot did not,
but it's a lesson learned in troubleshooting.  Before I hack through system
traces, I should try some quicker fixes first.  It would have been faster
had I just ripped it all out and rebuilt it.  It's hard to not get bogged
down sometimes when you're in the weeds.

Thanks again for the help.
Marty


On Mon, Jun 10, 2013 at 12:15 PM, Suresh Bhandari wrote:

> What happens when you make a call from HQ to BR1 phones? If phones ring,
> check if show sccp connections has expected output - that is, there is a
> reservation of 40K bandwidth for each ringing phones.
>
> You have partial configuration included here. Can you make sure that you
> have configured physical interfaces, Serial0/1/0 (from your config), on
> both routers to have ip rsvp bandwidth?
>
>
>
> On Mon, Jun 10, 2013 at 8:24 PM, Martin Sloan wrote:
>
>> Just re-read my configs, please note that BR1 *does* have the 'rsvp'
>> command under the dspfarm profile as well, it didn't copy over in my email:
>>
>>
>> dspfarm profile 1 mtp
>>  codec pass-through
>>  codec g729r8
>>  rsvp
>>  maximum sessions software 4
>>  associate application SCCP
>>
>>
>> On Mon, Jun 10, 2013 at 10:28 AM, Martin Sloan 
>> wrote:
>>
>>> I'm running into an issue getting RSVP to work on Vol 2 Lab 5 and would
>>> like some advice on what I might have missed.
>>>
>>> The requirement is to allow an equal number of calls (2) over redundant
>>> links from HQ->BR1 and 4 calls from HQ->BR2.  Also, RSVP should use 'video
>>> desired' allowing calls to proceed as audio only when there is not enough
>>> bandwidth for audio and video.  Just using HQ->BR1 as an example, so far I
>>> have configured:
>>>
>>> - bandwidth statements on Serial sub-interfaces:
>>>
>>> HQ:
>>>
>>> interface Serial0/1/0.1 point-to-point
>>>  bandwidth 384
>>>  ip address 10.10.111.1 255.255.255.252
>>>  ip ospf mtu-ignore
>>>  snmp trap link-status
>>>  frame-relay interface-dlci 201
>>>   class AutoQoS-FR-Se0/1/0-201
>>>   auto qos voip
>>>  ip rsvp bandwidth 64
>>> !
>>> interface Serial0/1/0.2 point-to-point
>>>  bandwidth 384
>>>  ip address 10.10.111.5 255.255.255.252
>>>  ip ospf mtu-ignore
>>>  snmp trap link-status
>>>  frame-relay interface-dlci 211
>>>   class AutoQoS-FR-Se0/1/0-201
>>>  ip rsvp bandwidth 64
>>>
>>> BR1:
>>>
>>> interface Serial0/1/0.1 point-to-point
>>>  bandwidth 384
>>>  ip address 10.10.111.2 255.255.255.252
>>>  ip ospf mtu-ignore
>>>  snmp trap link-status
>>>  frame-relay interface-dlci 101
>>>   class AutoQoS-FR-Se0/1/0-201
>>>  ip rsvp bandwidth 64
>>> !
>>> interface Serial0/1/0.2 point-to-point
>>>  bandwidth 384
>>>  ip address 10.10.111.6 255.255.255.252
>>>  ip ospf mtu-ignore
>>>  snmp trap link-status
>>>  frame-relay interface-dlci 111
>>>   class AutoQoS-FR-Se0/1/0-201
>>>  ip rsvp bandwidth 64
>>>
>>> - Software MTP on the router:
>>>
>>> HQ:
>>>
>>> sccp local Loopback0
>>> sccp ccm 10.10.210.10 identifier 2 priority 2 version 7.0
>>> sccp ccm 10.10.210.11 identifier 1 priority 1 version 7.0
>>> sccp
>>> !
>>> sccp ccm group 1
>>>  bind interface Loopback0
>>>  associate ccm 1 priority 1
>>>  associate ccm 2 priority 2
>>>  associate profile 1 register HQ-RSVP
>>> !
>>> dspfarm profile 1 mtp
>>>  codec pass-through
>>>  codec g729r8
>>>  rsvp
>>>  maximum sessions software 8
>>>  associate application SCCP
>>>
>>> BR1:
>>>
>>> sccp local Loopback0
>>> sccp ccm 10.10.210.11 identifier 1 priority 1 version 7.0
>>> sccp ccm 10.10.210.10 identifier 2 priority 2 version 7.0
>>> sccp
>>> !
>>> sccp ccm group 1
>>>  bind interface Loopback0
>>>  associate ccm 1 priority 1
>>>  associate ccm 2 priority 2
>>>  associate profile 1 register BR1-RSVP
>>> !
>>> dspfarm profile 1 mtp
>>>  codec pass-through
>>>  codec g729r8
>>>  maximum sessions software 4
>>>  associate application SCCP
>>>
>>> - Configured both MTP's in CUCM (both are registered)
>>> - Placed MTP's into their own MRG
>>> - Placed the RSVP MRG at the bottom of the MRGL for each site
>>> - Set Locations RSVP settings to mandatory (video desired)
>>> - Inter-region settings for HQ/BR1 is set to g729
>>> - HQ Device Pool contains HQ Location and Region
>>> - BR1 Device Pool contains BR1 Location and Region
>>> - HQ Device Pool is assigned to the phone placing the call to BR1
>>> - BR1 Device Pool is assigned to the phone I'm calling to from HQ
>>>
>>> With the above settings, I never see any RSVP messaging on the routers.
>>> I've done a debug sccp all and debug ip rsvp all and there is nothing sent
>>> in regard to RSVP CAC.  I've shut/no shut sccp about 100 times, rebooted
>>> the routers and rebooted the CUCM servers but still nothing.
>>>

Re: [OSL | CCIE_Voice] RSVP Problem

[Suresh Bhandari]

What happens when you make a call from HQ to BR1 phones? If phones ring,
check if show sccp connections has expected output - that is, there is a
reservation of 40K bandwidth for each ringing phones.

You have partial configuration included here. Can you make sure that you
have configured physical interfaces, Serial0/1/0 (from your config), on
both routers to have ip rsvp bandwidth?



On Mon, Jun 10, 2013 at 8:24 PM, Martin Sloan wrote:

> Just re-read my configs, please note that BR1 *does* have the 'rsvp'
> command under the dspfarm profile as well, it didn't copy over in my email:
>
>
> dspfarm profile 1 mtp
>  codec pass-through
>  codec g729r8
>  rsvp
>  maximum sessions software 4
>  associate application SCCP
>
>
> On Mon, Jun 10, 2013 at 10:28 AM, Martin Sloan wrote:
>
>> I'm running into an issue getting RSVP to work on Vol 2 Lab 5 and would
>> like some advice on what I might have missed.
>>
>> The requirement is to allow an equal number of calls (2) over redundant
>> links from HQ->BR1 and 4 calls from HQ->BR2.  Also, RSVP should use 'video
>> desired' allowing calls to proceed as audio only when there is not enough
>> bandwidth for audio and video.  Just using HQ->BR1 as an example, so far I
>> have configured:
>>
>> - bandwidth statements on Serial sub-interfaces:
>>
>> HQ:
>>
>> interface Serial0/1/0.1 point-to-point
>>  bandwidth 384
>>  ip address 10.10.111.1 255.255.255.252
>>  ip ospf mtu-ignore
>>  snmp trap link-status
>>  frame-relay interface-dlci 201
>>   class AutoQoS-FR-Se0/1/0-201
>>   auto qos voip
>>  ip rsvp bandwidth 64
>> !
>> interface Serial0/1/0.2 point-to-point
>>  bandwidth 384
>>  ip address 10.10.111.5 255.255.255.252
>>  ip ospf mtu-ignore
>>  snmp trap link-status
>>  frame-relay interface-dlci 211
>>   class AutoQoS-FR-Se0/1/0-201
>>  ip rsvp bandwidth 64
>>
>> BR1:
>>
>> interface Serial0/1/0.1 point-to-point
>>  bandwidth 384
>>  ip address 10.10.111.2 255.255.255.252
>>  ip ospf mtu-ignore
>>  snmp trap link-status
>>  frame-relay interface-dlci 101
>>   class AutoQoS-FR-Se0/1/0-201
>>  ip rsvp bandwidth 64
>> !
>> interface Serial0/1/0.2 point-to-point
>>  bandwidth 384
>>  ip address 10.10.111.6 255.255.255.252
>>  ip ospf mtu-ignore
>>  snmp trap link-status
>>  frame-relay interface-dlci 111
>>   class AutoQoS-FR-Se0/1/0-201
>>  ip rsvp bandwidth 64
>>
>> - Software MTP on the router:
>>
>> HQ:
>>
>> sccp local Loopback0
>> sccp ccm 10.10.210.10 identifier 2 priority 2 version 7.0
>> sccp ccm 10.10.210.11 identifier 1 priority 1 version 7.0
>> sccp
>> !
>> sccp ccm group 1
>>  bind interface Loopback0
>>  associate ccm 1 priority 1
>>  associate ccm 2 priority 2
>>  associate profile 1 register HQ-RSVP
>> !
>> dspfarm profile 1 mtp
>>  codec pass-through
>>  codec g729r8
>>  rsvp
>>  maximum sessions software 8
>>  associate application SCCP
>>
>> BR1:
>>
>> sccp local Loopback0
>> sccp ccm 10.10.210.11 identifier 1 priority 1 version 7.0
>> sccp ccm 10.10.210.10 identifier 2 priority 2 version 7.0
>> sccp
>> !
>> sccp ccm group 1
>>  bind interface Loopback0
>>  associate ccm 1 priority 1
>>  associate ccm 2 priority 2
>>  associate profile 1 register BR1-RSVP
>> !
>> dspfarm profile 1 mtp
>>  codec pass-through
>>  codec g729r8
>>  maximum sessions software 4
>>  associate application SCCP
>>
>> - Configured both MTP's in CUCM (both are registered)
>> - Placed MTP's into their own MRG
>> - Placed the RSVP MRG at the bottom of the MRGL for each site
>> - Set Locations RSVP settings to mandatory (video desired)
>> - Inter-region settings for HQ/BR1 is set to g729
>> - HQ Device Pool contains HQ Location and Region
>> - BR1 Device Pool contains BR1 Location and Region
>> - HQ Device Pool is assigned to the phone placing the call to BR1
>> - BR1 Device Pool is assigned to the phone I'm calling to from HQ
>>
>> With the above settings, I never see any RSVP messaging on the routers.
>> I've done a debug sccp all and debug ip rsvp all and there is nothing sent
>> in regard to RSVP CAC.  I've shut/no shut sccp about 100 times, rebooted
>> the routers and rebooted the CUCM servers but still nothing.
>>
>> I pulled some CUCM traces and I can see there is activity for RSVP but
>> it's unclear to me what the issue is.  Here are the last few lines, with an
>> 'SsCause' code that I haven't been able to dig up the meaning on:
>>
>> 000618017| 2013/06/07 11:26:50.937| 002| SdlSig|
>> SsUnregisterRelRejInterceptReq| tcc_intercept |
>> Cdcc(2,100,171,82)  | Cc(2,100,172,1) |
>> (2,100,37,1).114100-(SEP001E4A928455:10.10.200.72)| [R:NP - HP: 0, NP: 3,
>> LP: 0, VLP: 0, LZP: 0 DBP: 0]SsType=33554437 SsKey=0 SsNode=2
>> SsParty=45758182 handler=0
>> 000618018| 2013/06/07 11:26:50.937| 002| SdlSig|
>> SsUnregisterRelRejInterceptReq| tcc_intercept |
>> Cdcc(2,100,171,82)  | Cc(2,100,172,1) |
>> (2,100,37,1).114100-(SEP001E4A928455:10.10.200.72)| [R:NP - HP: 0, NP: 2,
>

[OSL | CCIE_Voice] Migration from CCIE Voice to Collaboration

[Kamran Ahsanullah]

Seems they have been listening, a voice CCIE can migration to
Collaboration CCIE by doing the written for Collaboration CCIE.

https://learningnetwork.cisco.com/docs/DOC-20968
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Re: [OSL | CCIE_Voice] RSVP Problem

[Martin Sloan]

Just re-read my configs, please note that BR1 *does* have the 'rsvp'
command under the dspfarm profile as well, it didn't copy over in my email:

dspfarm profile 1 mtp
 codec pass-through
 codec g729r8
 rsvp
 maximum sessions software 4
 associate application SCCP


On Mon, Jun 10, 2013 at 10:28 AM, Martin Sloan wrote:

> I'm running into an issue getting RSVP to work on Vol 2 Lab 5 and would
> like some advice on what I might have missed.
>
> The requirement is to allow an equal number of calls (2) over redundant
> links from HQ->BR1 and 4 calls from HQ->BR2.  Also, RSVP should use 'video
> desired' allowing calls to proceed as audio only when there is not enough
> bandwidth for audio and video.  Just using HQ->BR1 as an example, so far I
> have configured:
>
> - bandwidth statements on Serial sub-interfaces:
>
> HQ:
>
> interface Serial0/1/0.1 point-to-point
>  bandwidth 384
>  ip address 10.10.111.1 255.255.255.252
>  ip ospf mtu-ignore
>  snmp trap link-status
>  frame-relay interface-dlci 201
>   class AutoQoS-FR-Se0/1/0-201
>   auto qos voip
>  ip rsvp bandwidth 64
> !
> interface Serial0/1/0.2 point-to-point
>  bandwidth 384
>  ip address 10.10.111.5 255.255.255.252
>  ip ospf mtu-ignore
>  snmp trap link-status
>  frame-relay interface-dlci 211
>   class AutoQoS-FR-Se0/1/0-201
>  ip rsvp bandwidth 64
>
> BR1:
>
> interface Serial0/1/0.1 point-to-point
>  bandwidth 384
>  ip address 10.10.111.2 255.255.255.252
>  ip ospf mtu-ignore
>  snmp trap link-status
>  frame-relay interface-dlci 101
>   class AutoQoS-FR-Se0/1/0-201
>  ip rsvp bandwidth 64
> !
> interface Serial0/1/0.2 point-to-point
>  bandwidth 384
>  ip address 10.10.111.6 255.255.255.252
>  ip ospf mtu-ignore
>  snmp trap link-status
>  frame-relay interface-dlci 111
>   class AutoQoS-FR-Se0/1/0-201
>  ip rsvp bandwidth 64
>
> - Software MTP on the router:
>
> HQ:
>
> sccp local Loopback0
> sccp ccm 10.10.210.10 identifier 2 priority 2 version 7.0
> sccp ccm 10.10.210.11 identifier 1 priority 1 version 7.0
> sccp
> !
> sccp ccm group 1
>  bind interface Loopback0
>  associate ccm 1 priority 1
>  associate ccm 2 priority 2
>  associate profile 1 register HQ-RSVP
> !
> dspfarm profile 1 mtp
>  codec pass-through
>  codec g729r8
>  rsvp
>  maximum sessions software 8
>  associate application SCCP
>
> BR1:
>
> sccp local Loopback0
> sccp ccm 10.10.210.11 identifier 1 priority 1 version 7.0
> sccp ccm 10.10.210.10 identifier 2 priority 2 version 7.0
> sccp
> !
> sccp ccm group 1
>  bind interface Loopback0
>  associate ccm 1 priority 1
>  associate ccm 2 priority 2
>  associate profile 1 register BR1-RSVP
> !
> dspfarm profile 1 mtp
>  codec pass-through
>  codec g729r8
>  maximum sessions software 4
>  associate application SCCP
>
> - Configured both MTP's in CUCM (both are registered)
> - Placed MTP's into their own MRG
> - Placed the RSVP MRG at the bottom of the MRGL for each site
> - Set Locations RSVP settings to mandatory (video desired)
> - Inter-region settings for HQ/BR1 is set to g729
> - HQ Device Pool contains HQ Location and Region
> - BR1 Device Pool contains BR1 Location and Region
> - HQ Device Pool is assigned to the phone placing the call to BR1
> - BR1 Device Pool is assigned to the phone I'm calling to from HQ
>
> With the above settings, I never see any RSVP messaging on the routers.
> I've done a debug sccp all and debug ip rsvp all and there is nothing sent
> in regard to RSVP CAC.  I've shut/no shut sccp about 100 times, rebooted
> the routers and rebooted the CUCM servers but still nothing.
>
> I pulled some CUCM traces and I can see there is activity for RSVP but
> it's unclear to me what the issue is.  Here are the last few lines, with an
> 'SsCause' code that I haven't been able to dig up the meaning on:
>
> 000618017| 2013/06/07 11:26:50.937| 002| SdlSig|
> SsUnregisterRelRejInterceptReq| tcc_intercept |
> Cdcc(2,100,171,82)  | Cc(2,100,172,1) |
> (2,100,37,1).114100-(SEP001E4A928455:10.10.200.72)| [R:NP - HP: 0, NP: 3,
> LP: 0, VLP: 0, LZP: 0 DBP: 0]SsType=33554437 SsKey=0 SsNode=2
> SsParty=45758182 handler=0
> 000618018| 2013/06/07 11:26:50.937| 002| SdlSig|
> SsUnregisterRelRejInterceptReq| tcc_intercept |
> Cdcc(2,100,171,82)  | Cc(2,100,172,1) |
> (2,100,37,1).114100-(SEP001E4A928455:10.10.200.72)| [R:NP - HP: 0, NP: 2,
> LP: 0, VLP: 0, LZP: 0 DBP: 0]SsType=33554437 SsKey=0 SsNode=2
> SsParty=45758182 handler=0
> 000618019| 2013/06/07 11:26:50.937| 002| SdlSig|
> SsClearCallReq| tcc_intercept |
> Cdcc(2,100,171,82)  | Cc(2,100,172,1) |
> (2,100,37,1).114100-(SEP001E4A928455:10.10.200.72)| [R:NP - HP: 0, NP: 1,
> LP: 0, VLP: 0, LZP: 0 DBP: 0]SsType=33554437 SsKey=40 SsNode=2
> SsParty=45758182 *SsCause=125 *clearCallRequestor=0
> clearCallInstruction=1 FDataType=0opId=0invokeId=0resultExp=F
>
> Can someone take a look and let me 

[OSL | CCIE_Voice] RSVP Problem

[Martin Sloan]

I'm running into an issue getting RSVP to work on Vol 2 Lab 5 and would
like some advice on what I might have missed.

The requirement is to allow an equal number of calls (2) over redundant
links from HQ->BR1 and 4 calls from HQ->BR2.  Also, RSVP should use 'video
desired' allowing calls to proceed as audio only when there is not enough
bandwidth for audio and video.  Just using HQ->BR1 as an example, so far I
have configured:

- bandwidth statements on Serial sub-interfaces:

HQ:

interface Serial0/1/0.1 point-to-point
 bandwidth 384
 ip address 10.10.111.1 255.255.255.252
 ip ospf mtu-ignore
 snmp trap link-status
 frame-relay interface-dlci 201
  class AutoQoS-FR-Se0/1/0-201
  auto qos voip
 ip rsvp bandwidth 64
!
interface Serial0/1/0.2 point-to-point
 bandwidth 384
 ip address 10.10.111.5 255.255.255.252
 ip ospf mtu-ignore
 snmp trap link-status
 frame-relay interface-dlci 211
  class AutoQoS-FR-Se0/1/0-201
 ip rsvp bandwidth 64

BR1:

interface Serial0/1/0.1 point-to-point
 bandwidth 384
 ip address 10.10.111.2 255.255.255.252
 ip ospf mtu-ignore
 snmp trap link-status
 frame-relay interface-dlci 101
  class AutoQoS-FR-Se0/1/0-201
 ip rsvp bandwidth 64
!
interface Serial0/1/0.2 point-to-point
 bandwidth 384
 ip address 10.10.111.6 255.255.255.252
 ip ospf mtu-ignore
 snmp trap link-status
 frame-relay interface-dlci 111
  class AutoQoS-FR-Se0/1/0-201
 ip rsvp bandwidth 64

- Software MTP on the router:

HQ:

sccp local Loopback0
sccp ccm 10.10.210.10 identifier 2 priority 2 version 7.0
sccp ccm 10.10.210.11 identifier 1 priority 1 version 7.0
sccp
!
sccp ccm group 1
 bind interface Loopback0
 associate ccm 1 priority 1
 associate ccm 2 priority 2
 associate profile 1 register HQ-RSVP
!
dspfarm profile 1 mtp
 codec pass-through
 codec g729r8
 rsvp
 maximum sessions software 8
 associate application SCCP

BR1:

sccp local Loopback0
sccp ccm 10.10.210.11 identifier 1 priority 1 version 7.0
sccp ccm 10.10.210.10 identifier 2 priority 2 version 7.0
sccp
!
sccp ccm group 1
 bind interface Loopback0
 associate ccm 1 priority 1
 associate ccm 2 priority 2
 associate profile 1 register BR1-RSVP
!
dspfarm profile 1 mtp
 codec pass-through
 codec g729r8
 maximum sessions software 4
 associate application SCCP

- Configured both MTP's in CUCM (both are registered)
- Placed MTP's into their own MRG
- Placed the RSVP MRG at the bottom of the MRGL for each site
- Set Locations RSVP settings to mandatory (video desired)
- Inter-region settings for HQ/BR1 is set to g729
- HQ Device Pool contains HQ Location and Region
- BR1 Device Pool contains BR1 Location and Region
- HQ Device Pool is assigned to the phone placing the call to BR1
- BR1 Device Pool is assigned to the phone I'm calling to from HQ

With the above settings, I never see any RSVP messaging on the routers.
I've done a debug sccp all and debug ip rsvp all and there is nothing sent
in regard to RSVP CAC.  I've shut/no shut sccp about 100 times, rebooted
the routers and rebooted the CUCM servers but still nothing.

I pulled some CUCM traces and I can see there is activity for RSVP but it's
unclear to me what the issue is.  Here are the last few lines, with an
'SsCause' code that I haven't been able to dig up the meaning on:

000618017| 2013/06/07 11:26:50.937| 002| SdlSig|
SsUnregisterRelRejInterceptReq| tcc_intercept |
Cdcc(2,100,171,82)  | Cc(2,100,172,1) |
(2,100,37,1).114100-(SEP001E4A928455:10.10.200.72)| [R:NP - HP: 0, NP: 3,
LP: 0, VLP: 0, LZP: 0 DBP: 0]SsType=33554437 SsKey=0 SsNode=2
SsParty=45758182 handler=0
000618018| 2013/06/07 11:26:50.937| 002| SdlSig|
SsUnregisterRelRejInterceptReq| tcc_intercept |
Cdcc(2,100,171,82)  | Cc(2,100,172,1) |
(2,100,37,1).114100-(SEP001E4A928455:10.10.200.72)| [R:NP - HP: 0, NP: 2,
LP: 0, VLP: 0, LZP: 0 DBP: 0]SsType=33554437 SsKey=0 SsNode=2
SsParty=45758182 handler=0
000618019| 2013/06/07 11:26:50.937| 002| SdlSig|
SsClearCallReq| tcc_intercept |
Cdcc(2,100,171,82)  | Cc(2,100,172,1) |
(2,100,37,1).114100-(SEP001E4A928455:10.10.200.72)| [R:NP - HP: 0, NP: 1,
LP: 0, VLP: 0, LZP: 0 DBP: 0]SsType=33554437 SsKey=40 SsNode=2
SsParty=45758182 *SsCause=125 *clearCallRequestor=0 clearCallInstruction=1
FDataType=0opId=0invokeId=0resultExp=F

Can someone take a look and let me know if there's a glaring issue with my
configs?  I've set this up numerous times in the other labs so I'm either
blanking on the proper configs or missing a 'gotcha' somewhere.

Any help is appreciated.

Marty
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Re: [OSL | CCIE_Voice] [EXAM FAILED: Really Strange ]

[Daniel Pagan]

Sounds to me like the proctor was unable to run though his/her tests against 
your lab. Or, what I should say is, the proctor most likely had issues 
verifying step #1 and, because of that failure, never went to step #2, 3, etc. 
For example, if for some reason he/she cannot access one of the routers, 
there's no continuing on to verifying WAN QoS, gateway configs, etc., which 
would result in a 0% for those topics regardless of what was done.

That's the only logical explanation I could come up with. My suggestion is take 
a break for a day or two and then retrace your steps and redo the lab as Abel 
mentioned below.

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Abel ...
Sent: Monday, June 10, 2013 8:37 AM
To: ie ravindra
Cc: CCIE Study
Subject: Re: [OSL | CCIE_Voice] [EXAM FAILED: Really Strange ]


You're not first one to said the same history about the voice lab. My 
suggestion is try to lab down what you did in the lab, check your solution and 
try to remember if they really asked you that or if that is the usual solution 
provided by cisco site.

Abel Mateo
CCIE#28546
R/S | Voice
On Jun 9, 2013 11:36 AM, "ie ravindra" 
mailto:ieravin...@gmail.com>> wrote:
Dear All,
I got CCIE lab exam for the first time and I am really unsatisfied about the 
marks I got. unfortunately I was not able to complete my lab properly but I 
have configured few sections in good manner. Unfortunately I didn't knew that I 
removed a cable from a phone and replugged due to phones were not registered 
with CUCM correctly. when I got a result sheet it was really strange and they 
given me 0% for each and every section. So is there any way to appeal the exam 
to cisco and get another chance to attain this valuable certification. I know 
that I am technically competent. but now I am totally depressed.

Thanks,
Ravi,.

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Re: [OSL | CCIE_Voice] [EXAM FAILED: Really Strange ]

[Abel ...]

You're not first one to said the same history about the voice lab. My
suggestion is try to lab down what you did in the lab, check your solution
and try to remember if they really asked you that or if that is the usual
solution provided by cisco site.

Abel Mateo
CCIE#28546
R/S | Voice
On Jun 9, 2013 11:36 AM, "ie ravindra"  wrote:

> Dear All,
>
> I got CCIE lab exam for the first time and I am really unsatisfied about
> the marks I got. unfortunately I was not able to complete my lab properly
> but I have configured few sections in good manner. Unfortunately I didn't
> knew that I removed a cable from a phone and replugged due to phones were
> not registered with CUCM correctly. when I got a result sheet it was really
> strange and they given me 0% for each and every section. So is there any
> way to appeal the exam to cisco and get another chance to attain this
> valuable certification. I know that I am technically competent. but now I
> am totally depressed.
>
> Thanks,
> Ravi,.
>
> ___
> 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] Sip Troubleshooting

[Suresh Bhandari]

For me its the Content-Length:xxx with non-zero value.


On Fri, Jun 7, 2013 at 5:13 PM, ccie2k12  wrote:

>  I know the below message is clearly telling us that calling side is
> doing SIP Early offer and using TCP as the transport protocol
>
> but from the below message where do you think we should mark  for TCP
> and Early Offer
>
> ** **
>
> TCP 
>
> ** **
>
> 1) The first line "//SIP/SIPTcp...Outgoing SIP TCP"
>
> 2) ""
>
> ** **
>
> For Early Offer
>
> ** **
>
> 1)Content-Type: application/sdp
>
> 2)v=0
>
> ** **
>
> *CCM|//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to
> 151.1.2.2 on port 5060 index 6 *
>
> INVITE sip:911@151.1.2.2:5060 SIP/2.0
>
> Date: Fri, 07 Jun 2013 04:09:56 GMT
>
> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
> SUBSCRIBE, NOTIFY
>
> From: "HQPH1"  >;tag=06bb5c4b-f416-48f6-9225-822962b6cef1-29652495
>
> Allow-Events: presence, kpml
>
> P-Asserted-Identity: "HQPH1" 
>
> Supported: timer,resource-priority,replaces
>
> Min-SE:  1800
>
> Remote-Party-ID: "HQPH1"  >;party=calling;screen=yes;privacy=off
>
> Content-Length: 210
>
> User-Agent: Cisco-CUCM7.0
>
> To: 
>
> Contact: 
>
> Expires: 180
>
> Content-Type: application/sdp
>
> Call-ID: 1b618780-1b115d14-5-a0a01b1@151.1.10.10
>
> Via: SIP/2.0/TCP 151.1.10.10:5060;branch=z9hG4bK77edd7cbb
>
> CSeq: 101 INVITE
>
> Session-Expires:  1800
>
> Max-Forwards: 70
>
> ** **
>
> v=0
>
> o=CiscoSystemsCCM-SIP 2000 1 IN IP4 151.1.10.10
>
> s=SIP Call
>
> c=IN IP4 151.1.254.1
>
> t=0 0
>
> m=audio 16416 RTP/AVP 0 101
>
> a=rtpmap:0 PCMU/8000
>
> a=ptime:20
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-15
>
> ___
> 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
>



-- 
Suresh Bhandari
___
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