date:20111202

[OSL | CCIE_Voice] CCNA Voice certification preparation

2011-12-02 Thread 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

2011-12-02 Thread 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 aedamasc...@gmail.comwrote:

 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 mcl.nico...@gmail.comwrote:

  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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 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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 visit www.ipexpert.com

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

2011-12-02 Thread 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 baqari.voic...@gmail.com
Date: Fri, 2 Dec 2011 13:21:45 
To: Roy Vincent Aquinorva...@yahoo.com
Cc: ccie_voice@onlinestudylist.comccie_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 rva...@yahoo.com 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
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Re: [OSL | CCIE_Voice] [EMEA Cluster Gateway Dial Plan Assistance Munich Germany]

2011-12-02 Thread 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=indiacallto=germany


Regards,
Brajesh.


On Thu, Dec 1, 2011 at 7:23 PM,  michael.se...@compucom.com 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.














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 visit www.ipexpert.com

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 www.PlatinumPlacement.com
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Re: [OSL | CCIE_Voice] Introducing myself

2011-12-02 Thread 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 kf4...@gmail.com

 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 
 aedamasc...@gmail.comwrote:

 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 
 mcl.nico...@gmail.comwrote:

  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 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/





-- 
Nicolas MICHEL
Ingenieur Réseaux CCIE #29410
Tel: +33 6 08 72 75 97
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Re: [OSL | CCIE_Voice] Need Help with MoH

2011-12-02 Thread 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 datucha...@gmail.com
 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

 Gid  Lid Elog A/O FAX Tsec Codec   typePeer
 Address   IP Rip:udp

 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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[OSL | CCIE_Voice] Notify DTMF for CUCM

2011-12-02 Thread 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] DSPFarms for SRST

2011-12-02 Thread 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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Re: [OSL | CCIE_Voice] Need some help with GK/CUBE Remote Zone call debugging.

2011-12-02 Thread 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 ash.ayy...@gmail.com 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 ccielab...@gmail.com 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 FALSE
centralizedVideo FALSE
distributedVideo FALSE
  }
}
  }

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

2011-12-02 Thread 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 ccielab...@gmail.com 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 ash.ayy...@gmail.comwrote:

 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 ccielab...@gmail.com 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

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

2011-12-02 Thread datucha123 datucha123

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




On Fri, Dec 2, 2011 at 10:02 PM, ccielabrat ccielab...@gmail.com 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 ash.ayy...@gmail.comwrote:

 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 ccielab...@gmail.com 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

[OSL | CCIE_Voice] SRST MoH issue

2011-12-02 Thread 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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[OSL | CCIE_Voice] Automated Reply Re: CCIE_Voice Digest, Vol 70, Issue 14

2011-12-02 Thread 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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Re: [OSL | CCIE_Voice] DSPFarms for SRST

2011-12-02 Thread 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 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?

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[OSL | CCIE_Voice] Cisco 2811

2011-12-02 Thread 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.

2011-12-02 Thread 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 ccielab...@gmail.com 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 clm.c...@gmail.com 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 ccielab...@gmail.comwrote:

 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 ash.ayy...@gmail.comwrote:

 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 ccielab...@gmail.com
 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,

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

2011-12-02 Thread 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] DSPFarms for SRST

2011-12-02 Thread 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?

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

2011-12-02 Thread 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 ccielab...@gmail.com 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.


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Re: [OSL | CCIE_Voice] CCNA Voice certification preparation

2011-12-02 Thread 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:  mailto:mj...@ipexpert.com mj...@ipexpert.com

Office: (810) 326-1444 Ext: 331

Mobile +1 (810) 406-7218

www.ipexpert.com/catalog

 http://www.facebook.com/mattmjust Description: Description: facebook
http://twitter.com/#!/Matt_Just  http://www.linkedin.com/in/mattjust 

 

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] Need Help with MoH

2011-12-02 Thread 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 datucha...@gmail.com 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 datucha...@gmail.com 
 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
 
 Gid  Lid Elog A/O FAX Tsec Codec   typePeer Address 
   IP Rip:udp
 
 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] DSPFarms for SRST

2011-12-02 Thread 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 whl...@gmail.com 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?

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

2011-12-02 Thread 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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Re: [OSL | CCIE_Voice] DSPFarms for SRST

2011-12-02 Thread 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 kew...@gmail.com wrote: