date:20101007

Re: [OSL | CCIE_Voice] MoH to PSTN from SRST site

2010-10-07 Thread Mark Holloway

Hmm, PSTN to BR1 and IP to IP (inter and intra site) play multicast MoH piano 
music from route flash just fine, but for some reason when calling from BR1 to 
the PSTN and pressing HOLD on the BR1 phone it plays beep beep beep.  

Usually the issue is PSTN to IP because you need a voice class codec on the 
SUB/PUB dial peers that support G711, which I have, and PSTN to BR1 piano music 
streams multicast ok. Not sure what would cause IP to PSTN calls to fail 
streaming MoH and play beep beep beep.  Any ideas?



On Oct 7, 2010, at 1:36 PM, ayman labib wrote:

> Thanks for the reply. 
> 
> As it turns out.  Loopback interface is a required step.  Now everything is 
> working.  Thanks
> 
> Next challenge is to get Site HQ and SRST to use MoH with CME using the 
> Gatekeeper.  Thanks
> 
> From: ayman labib 
> To: amr thabt 
> Cc: ccie_voice@onlinestudylist.com
> Sent: Thu, October 7, 2010 3:49:41 PM
> Subject: Re: [OSL | CCIE_Voice] MoH to PSTN from SRST site
> 
> Thanks for the reply.
> 
> I do have the max ephone etc..  I removed my config to keep it short.
> I tried it with bind command and without.  Same Issue.
> I don't have Lo0 configured.  Everything is configured using the fa0/1 
> interface.  
> 
> Please have a look at the screen shots of my config.  I really appreciate 
> everyone's help.  2 days and it's driving me crazy. 
> 
> call-manager-fallback
>  secondary-dialtone 9
>  max-conferences 8 gain -6
>  transfer-system full-consult
>  ip source-address 192.168.31.10 port 2000 strict-match
>  max-ephones 10
>  max-dn 10
>  transfer-pattern .T
>  voicemail 912123945020
>  call-forward pattern .T
>  call-forward busy 12123945020
>  call-forward noan 12123945020 timeout 20
>  moh music-on-hold.au
>  multicast moh 239.1.1.1 port 16384 route 192.168.31.10
>  time-zone 8
> !
> 
> 
> From: amr thabt 
> To: ayman labib 
> Cc: ccie_voice@onlinestudylist.com
> Sent: Thu, October 7, 2010 3:07:59 PM
> Subject: Re: [OSL | CCIE_Voice] MoH to PSTN from SRST site
> 
> Hi Ayman,
> I have three comments that may help
> 1 Do you add max-dn and max-ephone under call-manager-fallback
> 2-in "ccm-manager music-on-hold bind fa0/1 " remove the bind use only 
> ccm-manager music-on-hold
> 3- in multicast command add both loopback and VLan SVI ip address.
>  
>  
> HTH
> AMR
> 
> 
> On Thu, Oct 7, 2010 at 9:56 PM, ayman labib  wrote:
> Just wondering if anyone encountered this problem.
> 
> I still can't get MOH when calling the PSTN phone and the site is not in SRST 
> mode.  According to the sh command below.  The call manager has done its job  
> but the GWY is not responding.  Any ideas?  MOH local and between HQ works 
> fine.  Just need a sanity check.  Thanks for all your help
> 
> SRST-Site#sh ccm-manager music-on-hold
> Current active multicast sessions : 1
>  Multicast   RTP port   Packets   Call   CodecIncoming
>  Address number in/outidInterface
> ===
> 239.1.1.1 16384   0/0  12   g711ulaw
> 
> ccm-manager music-on-hold bind fa0/1
> 
> call-manager-fallback
>  ip source-address 192.168.31.10 port 2000 strict-match
>  moh music-on-hold.au
>  multicast moh 239.1.1.1 port 16384 route 192.168.31.10
>  
> 
> http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_feature_guide09186a00802d1c31.html#wp1046789
> 
> From: ayman labib 
> To: ccie_voice@onlinestudylist.com
> Cc: ccie_voice@onlinestudylist.com
> Sent: Wed, October 6, 2010 9:45:12 AM
> Subject: MoH to PSTN from SRST site
> 
> 
> Hello Experts,
> 
> Follow up to Mark's email about Moh to PSTN.  I don't hear the Piano music as 
> well.  Inter-site and Intra-site with HQ works.  
> 
> I see the Muticast on the gateway is invoked and on the server, but don't 
> hear anything.  Any idea?  Thanks in advance
> 
> admin:show perf query class "Cisco MOH Device"
> ==>query class :
> 
>  - Perf class (Cisco MOH Device) has instances and values:
> MOH_2   -> MOHHighestActiveResources  = 1
> MOH_2   -> MOHMulticastResourceActive = 0
> MOH_2   -> MOHMulticastResourceAvailable  = 25
> MOH_2   -> MOHOutOfResources  = 0
> MOH_2   -> MOHTotalMulticastResources = 25
> MOH_2   -> MOHTotalUnicastResources   = 250
> MOH_2   -> MOHUnicastResourceActive   = 0
> MOH_2   -> MOHUnicastResourceAvailable= 250
> MOH_3   -> MOHHighestActiveResources  = 1
> MOH_3   -> MOHMulticastResourceActive = 1
> MOH_3   -> MOHMulticastResourceAvailable  = 24
> MOH_3   -> MOHOutOfResources  = 0
> MOH_3   -> MOHTotalMulticastResources = 25
> MOH_3   -> MOHTotalUnicastResources   = 250
> MOH_3   -> MOHUnicastResourceActive   = 0
> MOH_3   -> MOHUnicastResourceAvailable= 250
> 
> 
> 
>

[OSL | CCIE_Voice] Forwarding GK Calls to CUE

2010-10-07 Thread CCIE Voice GMAIL

Hey everyone,

 

I'm struggling to get calls to work from HQ or SB to SC through the GK to
CUE.  I'm figuring it's a problem with the Transcoder b/c the calls across
the WAN are G729 and CUE only accepts G711ulaw.

 

Any ideas what to do for this? 

 

Here is the relevant configurations:

 

 

 

<  VOICE SERVICE VOIP  >

 

voice service voip 

 allow-connections h323 to h323

 allow-connections h323 to sip

 allow-connections sip to h323

 allow-connections sip to sip

 no supplementary-service h225-notify cid-update

 fax protocol cisco 

 sip

  bind control source-interface Vlan250

  bind media source-interface Vlan250

  registrar server expires max 1500 min 300

 

 

<  DIAL PEER TO CUE  >

 

dial-peer voice 17 voip

 description TO CUE

 translation-profile outgoing TO_CUE

 destination-pattern 45[056][056]

 session protocol sipv2

 session target ipv4:10.5.202.254

 dtmf-relay sip-notify

 codec g711ulaw

 no vad

 

 

<  TRANSLATION TO CUE  >

 

voice translation-profile TO_CUE

 translate calling 14

 translate called 13

 

voice translation-rule 13

 rule 1 /^[234]...$/ /\0/

 

 

<  MEDIA RESOURCES  >

 

 

telephony-service

 sdspfarm units 3

 sdspfarm tag 1 SC_CONF

 sdspfarm tag 2 SC_MTP

 sdspfarm tag 3 SC_XCODE

 conference hardware

 

sccp local Vlan250

sccp ccm 10.5.202.1 identifier 1 version 7.0 

sccp

!

sccp ccm group 1

 associate ccm 1 priority 1

 associate profile 1 register SC_CONF

 associate profile 2 register SC_MTP

 associate profile 3 register SC_XCODE

!

dspfarm profile 3 transcode  

 codec g711ulaw

 codec g711alaw

 codec g729ar8

 codec g729abr8

 codec g729r8

 maximum sessions 3

 associate application SCCP

!

dspfarm profile 1 conference  

 codec g711ulaw

 codec g711alaw

 codec g729ar8

 codec g729abr8

 codec g729r8

 codec g729br8

 maximum sessions 1

 associate application SCCP

!

dspfarm profile 2 mtp  

 codec g729r8

 maximum sessions software 3

 associate application SCCP

 

<  EPHONE CONFIGURATION  >

 

ephone-dn  1  octo-line

 number 4001 no-reg primary

 label 4001

 description +442321314001

 name Site C Phone 1

 mobility

 snr 999 delay 2 timeout 10 cfwd-noan 4500

 allow watch

 call-forward busy 4500

 call-forward noan 4500 timeout 10

 

 

<  CUE SERVICE ENGINE   >

 

ip route 10.5.202.254 255.255.255.255 Service-Engine1/0

interface Service-Engine1/0

 ip unnumbered Vlan250

 service-module ip address 10.5.202.254 255.255.255.0

 service-module ip default-gateway 10.5.202.1

 

 

I was concerned that maybe the media resources didn't register, so I did a
"show sccp" command.  When doing this I found that the MTP wasn't
registering.  Does CME not support MTPs?  Or is it that I have a transcoder
registering as well?  I figure that the Transcoder should be able to handle
the calls from the GK to CUE as the transcoder is supporting both g711ulaw
and g729r8.

 

SCCP Admin State: UP

Gateway Local Interface: Vlan250

IPv4 Address: 10.5.202.1

Port Number: 2000

IP Precedence: 5

User Masked Codec list: None

Call Manager: 10.5.202.1, Port Number: 2000

Priority: N/A, Version: 7.0, Identifier: 1

Trustpoint: N/A

 

Conferencing Oper State: ACTIVE - Cause Code: NONE

Active Call Manager: 10.5.202.1, Port Number: 2000

TCP Link Status: CONNECTED, Profile Identifier: 1

Reported Max Streams: 8, Reported Max OOS Streams: 0

Supported Codec: g711ulaw, Maximum Packetization Period: 30

Supported Codec: g711alaw, Maximum Packetization Period: 30

Supported Codec: g729ar8, Maximum Packetization Period: 60

Supported Codec: g729abr8, Maximum Packetization Period: 60

Supported Codec: g729r8, Maximum Packetization Period: 60

Supported Codec: g729br8, Maximum Packetization Period: 60

Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30

Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30

Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization
Period: 30

  

Transcoding Oper State: ACTIVE - Cause Code: NONE

Active Call Manager: 10.5.202.1, Port Number: 2000

TCP Link Status: CONNECTED, Profile Identifier: 3

Reported Max Streams: 6, Reported Max OOS Streams: 0

Supported Codec: g711ulaw, Maximum Packetization Period: 30

Supported Codec: g711alaw, Maximum Packetization Period: 30

Supported Codec: g729ar8, Maximum Packetization Period: 60

Supported Codec: g729abr8, Maximum Packetization Period: 60

Supported Codec: g729r8, Maximum Packetization Period: 60

Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30

Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30

Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization
Period: 30

 

MTP Oper State: ACTIVE_IN_PROGRESS - Cause Code: CCM_REGISTER_FA

Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 56, Issue 73

2010-10-07 Thread Paul Kruger

Hey Amr,,
This is what i was trying to tell you earlier. You want to register the MTP
in CUCM, right? But look at your address in the 2nd line of your config:

"sccp ccm 10.10.200.3 identifier 1 version 5.0.1"

This is pointing to your GW, not CUCM. The  MTP needs to be as part of
another sccp ccm group. Try this:

sccp ccm x.x.x.x id 2 ver 7 (where x.x.x.x is the IP of your Primary
CPE/SUB)
sccp ccm x.x.x.y id 3 ver 7 (where x.x.x.y is the IP of your Secondary
CPE/PUB)
!
sccp ccm group 2
 associate ccm 2 priority 1
 associate ccm 3 priority 2
 associate profile 10 register hq-mtp

And then try and reg it again.

Let me know how it goes

On Thu, Oct 7, 2010 at 11:32 PM, Amr Sherif  wrote:

>  I created a software mtp on HQ-RTR ,create mtp on UCM (only had option for
> IOS ENHANCED MTP ) ,add it to a MRGL and applied to the GK device pool which
> is inserted in the GK trunk and check mtp required check mark but the MTP is
> not registered with UCM ,what am i missing here ?! is the codec right .
>
> *Here is my configuration:*
>
> sccp local FastEthernet0/0.20
> sccp ccm 10.10.200.3 identifier 1 version 5.0.1
> sccp
> !
> sccp ccm group 1
>  bind interface FastEthernet0/0.20
>  associate ccm 1 priority 1
>  associate profile 10 register hq-mtp
>  associate profile 1 register hq-xcode
>  signaling dscp af31
>
> dspfarm profile 1 transcode
>  codec g711ulaw
>  codec g711alaw
>  codec g729ar8
>  codec g729abr8
>  maximum sessions 3
>  associate application SCCP
> !
> dspfarm profile 10 mtp
>  codec g729r8
>  maximum sessions software 4
>  associate application SCCP
>
>
> Best regards,
>
>
>
> Amr Sherif
> Senior Network Voice Engineer
> CCNA,CCNP,CCVP and CCIE Voice Written *(Certified)*
> CCIE Voice Lab *(In Progress)*
> CEL: +966501462699
> Email: *miroale...@hotmail.com *
>
>
>
>
>
> >
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] RE : [OSL|VOICE] GK and fast start?

2010-10-07 Thread Romain Mullier

Yes it is what the problem was. Thank you all.

On Thu, Oct 7, 2010 at 6:11 PM, Friderich Claude wrote:

>   Hi
>
> Did you put your gk-trunk name in the service parameters of the CCM ??
> to use the port 1720 for this trunk name ...
> make a search for trunk  in service parameters
>
> Regards
>
>
> --
> *De:* ccie_voice-boun...@onlinestudylist.com de la part de Romain Mullier
> *Date:* jeu. 10/7/2010 9:43
> *À:* ccie_voice@onlinestudylist.com
> *Objet :* [OSL | CCIE_Voice] [OSL|VOICE] GK and fast start?
>
> All,
>
> In Volume 2 lab 2, I have been held up by the GK question 4.4. The output I
> have is
>
> HQ-RTR#sh gatekeeper endpoints
> GATEKEEPER ENDPOINT REGISTRATION
> 
> CallSignalAddr  Port  RASSignalAddr   Port  Zone Name TypeFlags
> --- - --- - - -
> 10.10.110.3 1720  10.10.110.3 57781 HQH323-GW
> H323-ID: BR2-RTR
> Voice Capacity Max.=  Avail.=  Current.= 0
> 10.10.210.1041115 10.10.210.1033212 HQVOIP-GW
> H323-ID: gk-trunk_1
> Voice Capacity Max.=  Avail.=  Current.= 0
> 10.10.210.1135023 10.10.210.1132786 HQVOIP-GW
> H323-ID: gk-trunk_2
> Voice Capacity Max.=  Avail.=  Current.= 0
> Total number of active registrations = 3
>
> The question asks that the port be 1720 for both CUCM instead of the random
> ports I have (41115 and 35023).
> I was thinking Fast Start might be involved in the solution but I couldn't
> figure it out.
> Has anyone struggled with the same issue?
>
> thanks
>
> --
> This email was Anti Virus checked by Astaro Security Gateway.
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

[OSL | CCIE_Voice] RE : CCIE_Voice Digest , Vol 56, Issue 73

2010-10-07 Thread Friderich Claude

you have to register the mtp to your call manager and not to the hq-rtr
 
just put this config 
 
sccp ccm 10.10.210.11 identifier 2 version 5.0.1
sccp ccm 10.10.210.10 identifier 3 version 5.0.1
 
sccp ccm group 2
associate ccm 2 priority 1
associate ccm 3 priority 2
associate profile 10 register hq-mtp
 
 
dspfarm profile 10 mtp  
 codec g729r8
 maximum sessions software 4
 associate application SCCP

no sccp
sccp

regards



De: ccie_voice-boun...@onlinestudylist.com de la part de Amr Sherif
Date: jeu. 10/7/2010 11:32
À: ccie_voice@onlinestudylist.com
Objet : Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 56, Issue 73


I created a software mtp on HQ-RTR ,create mtp on UCM (only had option for IOS 
ENHANCED MTP ) ,add it to a MRGL and applied to the GK device pool which is 
inserted in the GK trunk and check mtp required check mark but the MTP is not 
registered with UCM ,what am i missing here ?! is the codec right . 
 
Here is my configuration: 

sccp local FastEthernet0/0.20
sccp ccm 10.10.200.3 identifier 1 version 5.0.1 
sccp
!
sccp ccm group 1
 bind interface FastEthernet0/0.20
 associate ccm 1 priority 1
 associate profile 10 register hq-mtp
 associate profile 1 register hq-xcode
 signaling dscp af31
 
dspfarm profile 1 transcode  
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 maximum sessions 3
 associate application SCCP
!
dspfarm profile 10 mtp  
 codec g729r8
 maximum sessions software 4
 associate application SCCP


Best regards,





Amr Sherif
Senior Network Voice Engineer
CCNA,CCNP,CCVP and CCIE Voice Written (Certified)
CCIE Voice Lab (In Progress)
CEL: +966501462699
Email: miroale...@hotmail.com  





  
> From: ccie_voice-requ...@onlinestudylist.com
> Subject: CCIE_Voice Digest, Vol 56, Issue 73
> To: ccie_voice@onlinestudylist.com
> Date: Thu, 7 Oct 2010 11:02:13 -0400
> 
> Send CCIE_Voice mailing list submissions to
> ccie_voice@onlinestudylist.com
> 
> To subscribe or unsubscribe via the World Wide Web, visit
> http://onlinestudylist.com/mailman/listinfo/ccie_voice
> or, via email, send a message with subject or body 'help' to
> ccie_voice-requ...@onlinestudylist.com
> 
> You can reach the person managing the list at
> ccie_voice-ow...@onlinestudylist.com
> 
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of CCIE_Voice digest..."
> 
> 
> Today's Topics:
> 
> 1. Re: Volume 2 Lab 1 4.1 Call issue (Paul Kruger)
> 2. RE?: Volume 2 Lab 1 4.1 Call issue - xcoder (Friderich Claude)
> 3. Help (?mer ketene)
> 4. Re: Help (Steve Denney (stdenney))
> 
> 
> --
> 
> Message: 1
> Date: Thu, 7 Oct 2010 15:25:50 +0200
> From: Paul Kruger 
> To: Tam Nhu 
> Cc: ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] Volume 2 Lab 1 4.1 Call issue
> Message-ID:
> 
> Content-Type: text/plain; charset="iso-8859-1"
> 
> Yeah. What Tam said. Just to note: the MRGL used for your trunk should
> include the IOS MTP you configured, and this MUST be registered with CUCM.
> Keep this in mind, since the Xcoder would be registered to CUCME on HQ-RTR.
> 
> On Thu, Oct 7, 2010 at 2:14 PM, Tam Nhu  wrote:
> 
> > Hi Amr,
> >
> > since you are doing lab 1, and it has CUBE involved right. So make sure
> > the following need to check and configure
> >
> > Configure Xcoder and IOS MTP on HQ
> > On the trunk, make sure to
> >
> > - Checked MTP
> > - Unchecked 'Wait for H245 capacity set'
> > - Checked Enable Inbound Fast Start
> >
> > Hope that help,
> > TN.
> >
> >
> > ___
> > For more information regarding industry leading CCIE Lab training, please
> > visit www.ipexpert.com
> >
> >
> -- next part --
> An HTML attachment was scrubbed...
> URL: 
> 
> --
> 
> Message: 2
> Date: Thu, 7 Oct 2010 16:15:27 +0200
> From: "Friderich Claude" 
> To: "Amr Sherif" ,
> 
> Subject: [OSL | CCIE_Voice] RE?: Volume 2 Lab 1 4.1 Call issue -
> xcoder
> Message-ID:
> 
> Content-Type: text/plain; charset="iso-8859-1"
> 
> Another thing
> 
> Incoming call leg to CME is G711 (DP incoming) as resquested
> if default codec for the sip phone is registered on cme (default G.729), you 
> will need a xcoder on cme
> Or just put a codec g711ulaw on the voice register pool and it gonna work 
> withtout any xcoder on the cme
> 
> Regards
> 
> 
> 
> 
> De: ccie_voice-boun...@onlinestudylist.com de la part de Friderich Claude
> Date: jeu. 10/7/2010 3:15
> ?: Amr Sherif; ccie_voice@onlinestudylist.com
> Objet : Re: [OSL | CCIE_Voice] Volume 2 Lab 1 4.1 Call issue - xcoder
> 
> 
> 
> Hi,
> 
> 
> 
> Did you put a xcoder on hq-rtr ?
> 
> 
> 
> br2ph (CME) < hq-rtr(GK) <-- CCM
> 
> zone UCME G711 CUBE G729 zone UCM
> 
> 
> 
> As you can see you're involving a different codec for the inbound call leg 
> and outbound call

[OSL | CCIE_Voice] RE : [OSL|VOICE] GK and fast start?

2010-10-07 Thread Friderich Claude

Hi
 
Did you put your gk-trunk name in the service parameters of the CCM ?? 
to use the port 1720 for this trunk name ...
make a search for trunk  in service parameters
 
Regards




De: ccie_voice-boun...@onlinestudylist.com de la part de Romain Mullier
Date: jeu. 10/7/2010 9:43
À: ccie_voice@onlinestudylist.com
Objet : [OSL | CCIE_Voice] [OSL|VOICE] GK and fast start?


All,

In Volume 2 lab 2, I have been held up by the GK question 4.4. The output I 
have is

HQ-RTR#sh gatekeeper endpoints
GATEKEEPER ENDPOINT REGISTRATION

CallSignalAddr  Port  RASSignalAddr   Port  Zone Name TypeFlags
--- - --- - - -
10.10.110.3 1720  10.10.110.3 57781 HQH323-GW
H323-ID: BR2-RTR
Voice Capacity Max.=  Avail.=  Current.= 0
10.10.210.1041115 10.10.210.1033212 HQVOIP-GW
H323-ID: gk-trunk_1
Voice Capacity Max.=  Avail.=  Current.= 0
10.10.210.1135023 10.10.210.1132786 HQVOIP-GW
H323-ID: gk-trunk_2
Voice Capacity Max.=  Avail.=  Current.= 0
Total number of active registrations = 3

The question asks that the port be 1720 for both CUCM instead of the random 
ports I have (41115 and 35023).
I was thinking Fast Start might be involved in the solution but I couldn't 
figure it out.
Has anyone struggled with the same issue?

thanks

-- 
This email was Anti Virus checked by Astaro Security Gateway. 
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 56, Issue 73

2010-10-07 Thread Amr Sherif


I created a software mtp on HQ-RTR ,create mtp on UCM (only had option for IOS 
ENHANCED MTP ) ,add it to a MRGL and applied to the GK device pool which is 
inserted in the GK trunk and check mtp required check mark but the MTP is not 
registered with UCM ,what am i missing here ?! is the codec right . 
 
Here is my configuration: 

sccp local FastEthernet0/0.20
sccp ccm 10.10.200.3 identifier 1 version 5.0.1 
sccp
!
sccp ccm group 1
 bind interface FastEthernet0/0.20
 associate ccm 1 priority 1
 associate profile 10 register hq-mtp
 associate profile 1 register hq-xcode
 signaling dscp af31
 
dspfarm profile 1 transcode  
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 maximum sessions 3
 associate application SCCP
!
dspfarm profile 10 mtp  
 codec g729r8
 maximum sessions software 4
 associate application SCCP


Best regards,



Amr Sherif
Senior Network Voice Engineer
CCNA,CCNP,CCVP and CCIE Voice Written (Certified)
CCIE Voice Lab (In Progress)
CEL: +966501462699
Email: miroale...@hotmail.com



 

> From: ccie_voice-requ...@onlinestudylist.com
> Subject: CCIE_Voice Digest, Vol 56, Issue 73
> To: ccie_voice@onlinestudylist.com
> Date: Thu, 7 Oct 2010 11:02:13 -0400
> 
> Send CCIE_Voice mailing list submissions to
> ccie_voice@onlinestudylist.com
> 
> To subscribe or unsubscribe via the World Wide Web, visit
> http://onlinestudylist.com/mailman/listinfo/ccie_voice
> or, via email, send a message with subject or body 'help' to
> ccie_voice-requ...@onlinestudylist.com
> 
> You can reach the person managing the list at
> ccie_voice-ow...@onlinestudylist.com
> 
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of CCIE_Voice digest..."
> 
> 
> Today's Topics:
> 
> 1. Re: Volume 2 Lab 1 4.1 Call issue (Paul Kruger)
> 2. RE?: Volume 2 Lab 1 4.1 Call issue - xcoder (Friderich Claude)
> 3. Help (?mer ketene)
> 4. Re: Help (Steve Denney (stdenney))
> 
> 
> --
> 
> Message: 1
> Date: Thu, 7 Oct 2010 15:25:50 +0200
> From: Paul Kruger 
> To: Tam Nhu 
> Cc: ccie_voice@onlinestudylist.com
> Subject: Re: [OSL | CCIE_Voice] Volume 2 Lab 1 4.1 Call issue
> Message-ID:
> 
> Content-Type: text/plain; charset="iso-8859-1"
> 
> Yeah. What Tam said. Just to note: the MRGL used for your trunk should
> include the IOS MTP you configured, and this MUST be registered with CUCM.
> Keep this in mind, since the Xcoder would be registered to CUCME on HQ-RTR.
> 
> On Thu, Oct 7, 2010 at 2:14 PM, Tam Nhu  wrote:
> 
> > Hi Amr,
> >
> > since you are doing lab 1, and it has CUBE involved right. So make sure
> > the following need to check and configure
> >
> > Configure Xcoder and IOS MTP on HQ
> > On the trunk, make sure to
> >
> > - Checked MTP
> > - Unchecked 'Wait for H245 capacity set'
> > - Checked Enable Inbound Fast Start
> >
> > Hope that help,
> > TN.
> >
> >
> > ___
> > For more information regarding industry leading CCIE Lab training, please
> > visit www.ipexpert.com
> >
> >
> -- next part --
> An HTML attachment was scrubbed...
> URL: 
> 
> --
> 
> Message: 2
> Date: Thu, 7 Oct 2010 16:15:27 +0200
> From: "Friderich Claude" 
> To: "Amr Sherif" ,
> 
> Subject: [OSL | CCIE_Voice] RE?: Volume 2 Lab 1 4.1 Call issue -
> xcoder
> Message-ID:
> 
> Content-Type: text/plain; charset="iso-8859-1"
> 
> Another thing
> 
> Incoming call leg to CME is G711 (DP incoming) as resquested
> if default codec for the sip phone is registered on cme (default G.729), you 
> will need a xcoder on cme
> Or just put a codec g711ulaw on the voice register pool and it gonna work 
> withtout any xcoder on the cme
> 
> Regards
> 
> 
> 
> 
> De: ccie_voice-boun...@onlinestudylist.com de la part de Friderich Claude
> Date: jeu. 10/7/2010 3:15
> ?: Amr Sherif; ccie_voice@onlinestudylist.com
> Objet : Re: [OSL | CCIE_Voice] Volume 2 Lab 1 4.1 Call issue - xcoder
> 
> 
> 
> Hi,
> 
> 
> 
> Did you put a xcoder on hq-rtr ?
> 
> 
> 
> br2ph (CME) < hq-rtr(GK) <-- CCM
> 
> zone UCME G711 CUBE G729 zone UCM
> 
> 
> 
> As you can see you're involving a different codec for the inbound call leg 
> and outbound call leg at the hq-rtr
> 
> 
> 
> And CUBE doesn't support handshake TCS at all
> 
> So you have to disable 'Wait for H245 capacity set on cucm trunk as well
> 
> 
> 
> Hope this gonna help ...
> 
> 
> 
> Regards
> 
> 
> 
> 
> 
> Claude Friderich
> 
> PreSales Support
> 
> ccvp_voice_sm
>  
> 
> NETCORE PSF S.A.
> 
> 49 rue du Baerendall
> 
> B.P.65 L-8201 Mamer
> 
> T?l?phone: 31 33 80-407
> 
> Fax: 31 33 80 8-407
> 
> GSM: 621 303 616
> 
> E-mail: cfrider...@netcore.lu  
> 
> 
> 
> From: ccie

Re: [OSL | CCIE_Voice] MoH to PSTN from SRST site

2010-10-07 Thread ayman labib

Thanks for the reply. 

As it turns out.  Loopback interface is a required step.  Now everything is 
working.  Thanks

Next challenge is to get Site HQ and SRST to use MoH with CME using the 
Gatekeeper.  Thanks





From: ayman labib 
To: amr thabt 
Cc: ccie_voice@onlinestudylist.com
Sent: Thu, October 7, 2010 3:49:41 PM
Subject: Re: [OSL | CCIE_Voice] MoH to PSTN from SRST site


Thanks for the reply.

I do have the max ephone etc..  I removed my config to keep it short.
I tried it with bind command and without.  Same Issue.
I don't have Lo0 configured.  Everything is configured using the fa0/1 
interface.  


Please have a look at the screen shots of my config.  I really appreciate 
everyone's help.  2 days and it's driving me crazy. 


call-manager-fallback
 secondary-dialtone 9
 max-conferences 8 gain -6
 transfer-system full-consult
 ip source-address 192.168.31.10 port 2000 strict-match
 max-ephones 10
 max-dn 10
 transfer-pattern .T
 voicemail 912123945020
 call-forward pattern .T
 call-forward busy 12123945020
 call-forward  noan 12123945020 timeout 20
 moh music-on-hold.au
 multicast moh 239.1.1.1 port 16384 route 192.168.31.10
 time-zone 8
!






From: amr thabt 
To: ayman labib 
Cc: ccie_voice@onlinestudylist.com
Sent: Thu, October 7, 2010 3:07:59 PM
Subject: Re: [OSL | CCIE_Voice] MoH to PSTN from SRST site


Hi Ayman,
I have three comments that may help
1 Do you add max-dn and max-ephone under call-manager-fallback
2-in "ccm-manager music-on-hold bind fa0/1 " remove the bind use only 
ccm-manager music-on-hold
3- in multicast command add both loopback and VLan SVI ip address.
 
 
HTH
AMR



On Thu, Oct 7, 2010 at 9:56 PM, ayman labib  wrote:

Just wondering if anyone encountered this problem.
>
>I still can't get MOH when calling the PSTN phone and the site is not in SRST 
>mode.  According to the sh command below.  The call manager has done its job  
>but the GWY is not responding.  Any ideas?  MOH local and between HQ works 
>fine.  Just need a sanity check.  Thanks for all your help
>
>SRST-Site#sh ccm-manager music-on-hold
>Current active multicast sessions : 1
> Multicast   RTP port   Packets   Call   CodecIncoming
> Address number in/outidInterface
>===
>239.1.1.1 16384   0/0  12   g711ulaw
>
>ccm-manager music-on-hold bind fa0/1
>
>call-manager-fallback
> ip source-address 192.168.31.10 port 2000 strict-match
> moh music-on-hold.au
> multicast moh 239.1.1.1 port 16384 route 192.168.31.10
> 
>
>http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_feature_guide09186a00802d1c31.html#wp1046789
>
>
>
>
>

 From: ayman labib 
>To: ccie_voice@onlinestudylist.com
>Cc: ccie_voice@onlinestudylist.com
>Sent: Wed, October 6, 2010 9:45:12 AM
>Subject: MoH to PSTN from SRST site
>
>
>
>
>
>Hello Experts,
>
>Follow up to Mark's email about Moh to PSTN.  I don't hear the Piano music as 
>well.  Inter-site and Intra-site with HQ works.  
>
>
>I see the Muticast on the gateway is invoked and on the server, but don't hear 
>anything.  Any idea?  Thanks in advance
>
>admin:show perf query class "Cisco MOH Device"
>==>query class :
>
> - Perf class (Cisco MOH Device) has instances and values:
>MOH_2   -> MOHHighestActiveResources  = 1
>MOH_2   -> MOHMulticastResourceActive = 0
>MOH_2   -> MOHMulticastResourceAvailable  = 25
>MOH_2   -> MOHOutOfResources  = 0
>MOH_2   -> MOHTotalMulticastResources = 25
>MOH_2   -> MOHTotalUnicastResources   = 250
>MOH_2   -> MOHUnicastResourceActive   = 0
>MOH_2   -> MOHUnicastResourceAvailable= 250
>MOH_3   -> MOHHighestActiveResources  = 1
>MOH_3   -> MOHMulticastResourceActive = 1
>MOH_3   -> MOHMulticastResourceAvailable  = 24
>MOH_3   -> MOHOutOfResources  = 0
>MOH_3   -> MOHTotalMulticastResources = 25
>MOH_3   -> MOHTotalUnicastResources   = 250
>MOH_3   -> MOHUnicastResourceActive   = 0
>MOH_3   -> MOHUnicastResourceAvailable= 250
>
>
>
>
>
>
>___
>For more information regarding industry leading CCIE Lab training, please 
>visit 
>www.ipexpert.com
>
>

___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

[OSL | CCIE_Voice] Is this CUE's normal Behaviour

2010-10-07 Thread Pithog Oil


Hi Expert 
please i will like to know if it is a normal behaviour for CUE to tell me, to 
record a message that is at least 2 seconds long , and even when i record a 
message longer than 2 seconds it continues to tell me the same thing.
i might have mis configured some parameters, kindly light me up on this concern.
Pithog oil


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[OSL | CCIE_Voice] your gatek issues

2010-10-07 Thread Pithog Oil

Hi romain
Go to service parameter search for the value "1720" then type in you gatekeeper 
registered name "gk-trunk" into the parameter allocated space, that will get 
the trick done for you.
That is a good way to harcode your port numbers.



  ___
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Re: [OSL | CCIE_Voice] [OSL|VOICE] GK and fast start?

2010-10-07 Thread Steve Denney (stdenney)

Nothing to do with Fast Start. You need to go to CCM service parameters,
and enter "gk-trunk" under "Device name of GK-controlled Trunk that will
use port 1720"

 

cheers, sd

 

 

From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Romain
Mullier
Sent: Thursday, October 07, 2010 3:44 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] [OSL|VOICE] GK and fast start?

 

All,

In Volume 2 lab 2, I have been held up by the GK question 4.4. The
output I have is

HQ-RTR#sh gatekeeper endpoints
GATEKEEPER ENDPOINT REGISTRATION

CallSignalAddr  Port  RASSignalAddr   Port  Zone Name Type
Flags
--- - --- - - 
-
10.10.110.3 1720  10.10.110.3 57781 HQH323-GW
H323-ID: BR2-RTR
Voice Capacity Max.=  Avail.=  Current.= 0
10.10.210.1041115 10.10.210.1033212 HQVOIP-GW
H323-ID: gk-trunk_1
Voice Capacity Max.=  Avail.=  Current.= 0
10.10.210.1135023 10.10.210.1132786 HQVOIP-GW
H323-ID: gk-trunk_2
Voice Capacity Max.=  Avail.=  Current.= 0
Total number of active registrations = 3

The question asks that the port be 1720 for both CUCM instead of the
random ports I have (41115 and 35023).
I was thinking Fast Start might be involved in the solution but I
couldn't figure it out.
Has anyone struggled with the same issue?

thanks

___
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[OSL | CCIE_Voice] [OSL|VOICE] GK and fast start?

2010-10-07 Thread Romain Mullier

All,

In Volume 2 lab 2, I have been held up by the GK question 4.4. The output I
have is

HQ-RTR#sh gatekeeper endpoints
GATEKEEPER ENDPOINT REGISTRATION

CallSignalAddr  Port  RASSignalAddr   Port  Zone Name TypeFlags
--- - --- - - -
10.10.110.3 1720  10.10.110.3 57781 HQH323-GW
H323-ID: BR2-RTR
Voice Capacity Max.=  Avail.=  Current.= 0
10.10.210.1041115 10.10.210.1033212 HQVOIP-GW
H323-ID: gk-trunk_1
Voice Capacity Max.=  Avail.=  Current.= 0
10.10.210.1135023 10.10.210.1132786 HQVOIP-GW
H323-ID: gk-trunk_2
Voice Capacity Max.=  Avail.=  Current.= 0
Total number of active registrations = 3

The question asks that the port be 1720 for both CUCM instead of the random
ports I have (41115 and 35023).
I was thinking Fast Start might be involved in the solution but I couldn't
figure it out.
Has anyone struggled with the same issue?

thanks
___
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www.ipexpert.com

Re: [OSL | CCIE_Voice] MoH to PSTN from SRST site

2010-10-07 Thread amr thabt

Hi Ayman,
I have three comments that may help
1 Do you add max-dn and max-ephone under call-manager-fallback
2-in "ccm-manager music-on-hold bind fa0/1 " remove the bind use only
ccm-manager music-on-hold
3- in multicast command add both loopback and VLan SVI ip address.


HTH
AMR


On Thu, Oct 7, 2010 at 9:56 PM, ayman labib  wrote:

>  Just wondering if anyone encountered this problem.
>
> I still can't get MOH when calling the PSTN phone and the site is not in
> SRST mode.  According to the sh command below.  The call manager has done
> its job  but the GWY is not responding.  Any ideas?  MOH local and between
> HQ works fine.  Just need a sanity check.  Thanks for all your help
>
> SRST-Site#sh ccm-manager music-on-hold
> Current active multicast sessions : 1
>  Multicast   RTP port   Packets   Call   CodecIncoming
>  Address number in/outid
> Interface
> ===
> 239.1.1.1 16384   0/0  12   g711ulaw
>
> ccm-manager music-on-hold bind fa0/1
>
> call-manager-fallback
>  ip source-address 192.168.31.10 port 2000 strict-match
>  moh music-on-hold.au
>  multicast moh 239.1.1.1 port 16384 route 192.168.31.10
>
>
>
> http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_feature_guide09186a00802d1c31.html#wp1046789
>
>  --
> *From:* ayman labib 
> *To:* ccie_voice@onlinestudylist.com
> *Cc:* ccie_voice@onlinestudylist.com
> *Sent:* Wed, October 6, 2010 9:45:12 AM
> *Subject:* MoH to PSTN from SRST site
>
>
> Hello Experts,
>
> Follow up to Mark's email about Moh to PSTN.  I don't hear the Piano music
> as well.  Inter-site and Intra-site with HQ works.
>
> I see the Muticast on the gateway is invoked and on the server, but don't
> hear anything.  Any idea?  Thanks in advance
>
> admin:show perf query class "Cisco MOH Device"
> ==>query class :
>
>  - Perf class (Cisco MOH Device) has instances and values:
> MOH_2   -> MOHHighestActiveResources  = 1
> MOH_2   -> MOHMulticastResourceActive = 0
> MOH_2   -> MOHMulticastResourceAvailable  = 25
> MOH_2   -> MOHOutOfResources  = 0
> MOH_2   -> MOHTotalMulticastResources = 25
> MOH_2   -> MOHTotalUnicastResources   = 250
> MOH_2   -> MOHUnicastResourceActive   = 0
> MOH_2   -> MOHUnicastResourceAvailable= 250
> MOH_3   -> MOHHighestActiveResources  = 1
> MOH_3   -> MOHMulticastResourceActive = 1
> MOH_3   -> MOHMulticastResourceAvailable  = 24
> MOH_3   -> MOHOutOfResources  = 0
> MOH_3   -> MOHTotalMulticastResources = 25
> MOH_3   -> MOHTotalUnicastResources   = 250
> MOH_3   -> MOHUnicastResourceActive   = 0
> MOH_3   -> MOHUnicastResourceAvailable= 250
>
>
>
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] MVA Troubleshooting lab 6 question 5.3

2010-10-07 Thread amr thabt

Hi Stutz,
 1- add translation rule& profile to dial-p 1997 to change the calling
number to be  '8884343' .
 2- if still have a problem , check css of RDP and may restart Mobile Voice
Service
 I hpoe this may help
HTH
AMR


On Thu, Oct 7, 2010 at 9:26 PM, Stutz, Bernhard  wrote:

>  Hi,
>
> I run into the same issue.
> furthermore i have to hairpin the call through a h323 gateway as all
> incoming calls come per mgcp to the callmanager. You have then to add a
> H.323 gateway to the same mgcp gateway which is possible.
>
> I got following dial--peers configured:
>
> dial-peer voice 1999 voip
>  service cmm
>  incoming called-number 1999
>  dtmf-relay h245-alphanumeric
>  codec g711ulaw
> !
> dial-peer voice 101 voip
>  preference 1
>  destination-pattern 1997
>  voice-class h323 1
>  session target ipv4:10.10.210.10
>  dtmf-relay h245-alphanumeric
>  codec g711ulaw
>  no vad
> Under callmanager i have 1997 as MVA Number defined at Media
> Ressources->Mobile Voice Access and also at service parameter
>
> When i call the mva the call comes in via mgcp, on ccm i have a route
> pattern that sends 1999 back to the h.323 configured gateway, then the
> service gets invoked. so far so good.
>
> I have remote destination configured with 8884343 and the call comes in as
> following:
>
> Oct  7 21:41:58.277: ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8  callref =
> 0x00B4
> Bearer Capability i = 0x8090A2
> Standard = CCITT
> Transfer Capability = Speech
> Transfer Mode = Circuit
> Transfer Rate = 64 kbit/s
> Channel ID i = 0xA98381
> Exclusive, Channel 1
> Progress Ind i = 0x8583 - Origination address is non-ISDN
> Calling Party Number i = 0x4180,
> Plan:ISDN, Type:Subscriber(local)
> Called Party Number i = 0xA1, '4158881999'
> Plan:ISDN, Type:National
> Oct  7 21:41:58.317: //-1/80DCADB41800/DPM/dpAssociateIncomingPeerCore:
>Calling Number=8884343, Called Number=1999, Voice-Interface=0x0,
>Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
> Type=PEER_TYPE_VOICE,
>Peer Info Type=DIALPEER_INFO_SPEECH
> Oct  7 21:41:58.317: //-1/80DCADB41800/DPM/dpAssociateIncomingPeerCore:
>Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1999
> Oct  7 21:41:58.317: //-1/80DCADB41800/DPM/dpAssociateIncomingPeerCore:
>Calling Number=8884343, Called Number=1999, Voice-Interface=0x0,
>Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
> Type=PEER_TYPE_VOICE,
>Peer Info Type=DIALPEER_INFO_SPEECH
> Oct  7 21:41:58.317: //-1/80DCADB41800/DPM/dpAssociateIncomingPeerCore:
> BR1-RTR#Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming
> Dial-peer=1999
> Oct  7 21:41:58.385: ISDN Se0/0/0:23 Q931: TX -> CALL_PROC pd = 8  callref
> = 0x80B4
> Channel ID i = 0xA98381
> Exclusive, Channel 1
> Oct  7 21:41:58.393: ISDN Se0/0/0:23 Q931: TX -> CONNECT pd = 8  callref =
> 0x80B4
> Oct  7 21:41:58.401: ISDN Se0/0/0:23 Q931: RX <- CONNECT_ACK pd = 8
> callref = 0x00B4
>
> Then i am getting asked for the pin which is been accepted. after that i
> choose option 1 and push 5002#
> Then the call gets disconnected:
>
> Oct  7 21:42:21.297: //-1//DPM/dpMatchPeersCore:
>Calling Number=, Called Number=1997, Peer Info Type=DIALPEER_INFO_SPEECH
> Oct  7 21:42:21.301: //-1//DPM/dpMatchPeersCore:
>Match Rule=DP_MATCH_DEST; Called Number=1997
> Oct  7 21:42:21.301: //-1//DPM/dpMatchPeersCore:
>Result=Success(0) after DP_MATCH_DEST
> Oct  7 21:42:21.301: //-1//DPM/dpMatchPeersMoreArg:
>Result=SUCCESS(0)
>List of Matched Outgoing Dial-peer(s):
>  1: Dial-peer Tag=101
> Oct  7 21:42:21.301: //-1/80DCADB41800/DPM/dpMatchPeersCore:
>Calling Number=, Called Number=1997, Peer Info Type=DIALPEER_INFO_SPEECH
> Oct  7 21:42:21.301: //-1/80DCADB41800/DPM/dpMatchPeersCore:
>Match Rule=DP_MATCH_DEST; Called Number=1997
> Oct  7 21:42:21.301: //-1/80DCADB41800/DPM/dpMatchPeersCore:
>Result=Success(0) after DP_MATCH_DEST
> Oct  7 21:42:21.301: //-1/80DCADB41800/DPM/dpMatchPeersMoreArg:
>Result=SUCCESS(0)
>List of Matched Outgoing Dial-peer(s):
>  1: Dial-peer Tag=101
> Oct  7 21:42:21.345: //-1//DPM/dpMatchPeersCore:
>Calling Number=, Called Number=8884343, Peer Info
> Type=DIALPEER_INFO_SPEECH
> Oct  7 21:42:21.345: //-1//DPM/dpMatchPeersCore:
>Match Rule=DP_MATCH_DEST; Called Number=8884343
> Oct  7 21:42:21.345: //-1//DPM/dpMatchPeersCore:
>No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)
> Oct  7 21:42:21.345: //-1//DPM/dpMatchPeers:
>Result=NO_MATCH(-1)
> Oct  7 21:42:21.345: //-1//DPM/dpMatchPeersCore:
>Calling Number=, Called Number=1997, Peer Info Type=DIALPEER_INFO_SPEECH
> Oct  7 21:42:21.345: //-1//DPM/dpMatchPeersCore:
>Match Rule=DP_MATCH_DEST; Called

Re: [OSL | CCIE_Voice] MoH to PSTN from SRST site

2010-10-07 Thread Wilson.Samuel

Hi Ayman,

I would prefer to have a look on the Regions and Codecs, I have had a similar 
issue like this
Regards
Wilson Samuel


From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ayman labib
Sent: Thursday, October 07, 2010 2:56 PM
To: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] MoH to PSTN from SRST site

Just wondering if anyone encountered this problem.

I still can't get MOH when calling the PSTN phone and the site is not in SRST 
mode.  According to the sh command below.  The call manager has done its job  
but the GWY is not responding.  Any ideas?  MOH local and between HQ works 
fine.  Just need a sanity check.  Thanks for all your help

SRST-Site#sh ccm-manager music-on-hold
Current active multicast sessions : 1
 Multicast   RTP port   Packets   Call   CodecIncoming
 Address number in/outidInterface
===
239.1.1.1 16384   0/0  12   g711ulaw

ccm-manager music-on-hold bind fa0/1

call-manager-fallback
 ip source-address 192.168.31.10 port 2000 strict-match
 moh music-on-hold.au
 multicast moh 239.1.1.1 port 16384 route 192.168.31.10


http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_feature_guide09186a00802d1c31.html#wp1046789


From: ayman labib 
To: ccie_voice@onlinestudylist.com
Cc: ccie_voice@onlinestudylist.com
Sent: Wed, October 6, 2010 9:45:12 AM
Subject: MoH to PSTN from SRST site

Hello Experts,

Follow up to Mark's email about Moh to PSTN.  I don't hear the Piano music as 
well.  Inter-site and Intra-site with HQ works.

I see the Muticast on the gateway is invoked and on the server, but don't hear 
anything.  Any idea?  Thanks in advance

admin:show perf query class "Cisco MOH Device"
==>query class :

 - Perf class (Cisco MOH Device) has instances and values:
MOH_2   -> MOHHighestActiveResources  = 1
MOH_2   -> MOHMulticastResourceActive = 0
MOH_2   -> MOHMulticastResourceAvailable  = 25
MOH_2   -> MOHOutOfResources  = 0
MOH_2   -> MOHTotalMulticastResources = 25
MOH_2   -> MOHTotalUnicastResources   = 250
MOH_2   -> MOHUnicastResourceActive   = 0
MOH_2   -> MOHUnicastResourceAvailable= 250
MOH_3   -> MOHHighestActiveResources  = 1
MOH_3   -> MOHMulticastResourceActive = 1
MOH_3   -> MOHMulticastResourceAvailable  = 24
MOH_3   -> MOHOutOfResources  = 0
MOH_3   -> MOHTotalMulticastResources = 25
MOH_3   -> MOHTotalUnicastResources   = 250
MOH_3   -> MOHUnicastResourceActive   = 0
MOH_3   -> MOHUnicastResourceAvailable= 250




___
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Re: [OSL | CCIE_Voice] MoH to PSTN from SRST site

2010-10-07 Thread ayman labib

Just wondering if anyone encountered this problem.

I still can't get MOH when calling the PSTN phone and the site is not in SRST 
mode.  According to the sh command below.  The call manager has done its job  
but the GWY is not responding.  Any ideas?  MOH local and between HQ works 
fine.  Just need a sanity check.  Thanks for all your help

SRST-Site#sh ccm-manager music-on-hold
Current active multicast sessions : 1
 Multicast   RTP port   Packets   Call   CodecIncoming
 Address number in/outidInterface
===
239.1.1.1 16384   0/0  12   g711ulaw

ccm-manager music-on-hold bind fa0/1

call-manager-fallback
 ip source-address 192.168.31.10 port 2000 strict-match
 moh music-on-hold.au
 multicast moh 239.1.1.1 port 16384 route 192.168.31.10
 

http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_feature_guide09186a00802d1c31.html#wp1046789






From: ayman labib 
To: ccie_voice@onlinestudylist.com
Cc: ccie_voice@onlinestudylist.com
Sent: Wed, October 6, 2010 9:45:12 AM
Subject: MoH to PSTN from SRST site




Hello Experts,

Follow up to Mark's email about Moh to PSTN.  I don't hear the Piano music as 
well.  Inter-site and Intra-site with HQ works.  


I see the Muticast on the gateway is invoked and on the server, but don't hear 
anything.  Any idea?  Thanks in advance

admin:show perf query class "Cisco MOH Device"
==>query class :

 - Perf class (Cisco MOH Device) has instances and values:
MOH_2   -> MOHHighestActiveResources  = 1
MOH_2   ->  MOHMulticastResourceActive = 0
MOH_2   -> MOHMulticastResourceAvailable  = 25
MOH_2   -> MOHOutOfResources  = 0
MOH_2   -> MOHTotalMulticastResources = 25
MOH_2   -> MOHTotalUnicastResources   = 250
MOH_2   -> MOHUnicastResourceActive   = 0
MOH_2   ->  MOHUnicastResourceAvailable= 250
MOH_3   -> MOHHighestActiveResources  = 1
MOH_3   -> MOHMulticastResourceActive = 1
MOH_3   -> MOHMulticastResourceAvailable  = 24
MOH_3   -> MOHOutOfResources  = 0
MOH_3   -> MOHTotalMulticastResources = 25
MOH_3   ->  MOHTotalUnicastResources   = 250
MOH_3   -> MOHUnicastResourceActive   = 0
MOH_3   -> MOHUnicastResourceAvailable= 250

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Re: [OSL | CCIE_Voice] HW conference with CAC problem

2010-10-07 Thread Roig Borrell, Francesc Xavier

Hi Prashant,

They aren't shared lines
Hq ph1 , 2001
Br1 ph1, 3001
Br2 ph2, 3002


De: Prashant Patel [mailto:prashantpatel...@gmail.com]
Enviado el: jueves, 07 de octubre de 2010 20:20
Para: Roig Borrell, Francesc Xavier
CC: ccie_voice@onlinestudylist.com
Asunto: Re: [OSL | CCIE_Voice] HW conference with CAC problem

Hi,

Do you have any shared lines in your scenario across the two phones you are 
conferencing ie shared lines between HQ , A or B phones? If so every call to 
shared line needs 24kbps across the WAN.

HTH
Prashant
On Thu, Oct 7, 2010 at 2:16 PM, Roig Borrell, Francesc Xavier 
mailto:francesc.ro...@tecnocom.es>> wrote:
Hi all,

I have this scenario

Hardware CFB configured  in HQ
Ad-hoc conferene started by hq phone1 with br1 phone1 and br1 phone2. No 
problem with this

HQ#SH sccp connections
sess_idconn_idstype mode codec   ripaddr rport sport
33556436   33554594   conf  sendrecv g729b   192.168.21.327628 17266
33556436   33554590   conf  sendrecv g711u   192.168.20.427408 16608
33556436   33554588   conf  sendrecv g729b   192.168.20.227820 18172
Total number of active session(s) 1, and connection(s) 3

The problems starts if I configure CAC in BR1
With br1 Location=48 , the  conference fails  (Message in hq phone1 Cannot 
complete the conference)
I have to configure at least Location=72 in order to make it work.
I have two g729 calls so with 48 should work. Any ideas why I need to configure 
72?

Thanks in advance!!
Xavi


___
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___
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Re: [OSL | CCIE_Voice] MVA Troubleshooting lab 6 question 5.3

2010-10-07 Thread Stutz, Bernhard

Hi,
 
I run into the same issue.
furthermore i have to hairpin the call through a h323 gateway as all incoming 
calls come per mgcp to the callmanager. You have then to add a H.323 gateway to 
the same mgcp gateway which is possible.
 
I got following dial--peers configured:
 
dial-peer voice 1999 voip
 service cmm
 incoming called-number 1999
 dtmf-relay h245-alphanumeric
 codec g711ulaw
!
dial-peer voice 101 voip
 preference 1
 destination-pattern 1997
 voice-class h323 1
 session target ipv4:10.10.210.10
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad

Under callmanager i have 1997 as MVA Number defined at Media Ressources->Mobile 
Voice Access and also at service parameter
 
When i call the mva the call comes in via mgcp, on ccm i have a route pattern 
that sends 1999 back to the h.323 configured gateway, then the service gets 
invoked. so far so good.
 
I have remote destination configured with 8884343 and the call comes in as 
following:
 
Oct  7 21:41:58.277: ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8  callref = 0x00B4
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98381
Exclusive, Channel 1
Progress Ind i = 0x8583 - Origination address is non-ISDN
Calling Party Number i = 0x4180, '8884343'
Plan:ISDN, Type:Subscriber(local)
Called Party Number i = 0xA1, '4158881999'
Plan:ISDN, Type:National
Oct  7 21:41:58.317: //-1/80DCADB41800/DPM/dpAssociateIncomingPeerCore:
   Calling Number=8884343, Called Number=1999, Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
Oct  7 21:41:58.317: //-1/80DCADB41800/DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1999
Oct  7 21:41:58.317: //-1/80DCADB41800/DPM/dpAssociateIncomingPeerCore:
   Calling Number=8884343, Called Number=1999, Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
Oct  7 21:41:58.317: //-1/80DCADB41800/DPM/dpAssociateIncomingPeerCore:
BR1-RTR#Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1999
Oct  7 21:41:58.385: ISDN Se0/0/0:23 Q931: TX -> CALL_PROC pd = 8  callref = 
0x80B4
Channel ID i = 0xA98381
Exclusive, Channel 1
Oct  7 21:41:58.393: ISDN Se0/0/0:23 Q931: TX -> CONNECT pd = 8  callref = 
0x80B4
Oct  7 21:41:58.401: ISDN Se0/0/0:23 Q931: RX <- CONNECT_ACK pd = 8  callref = 
0x00B4

 
Then i am getting asked for the pin which is been accepted. after that i choose 
option 1 and push 5002#
Then the call gets disconnected:
 
Oct  7 21:42:21.297: //-1//DPM/dpMatchPeersCore:
   Calling Number=, Called Number=1997, Peer Info Type=DIALPEER_INFO_SPEECH
Oct  7 21:42:21.301: //-1//DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=1997
Oct  7 21:42:21.301: //-1//DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
Oct  7 21:42:21.301: //-1//DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
 1: Dial-peer Tag=101
Oct  7 21:42:21.301: //-1/80DCADB41800/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=1997, Peer Info Type=DIALPEER_INFO_SPEECH
Oct  7 21:42:21.301: //-1/80DCADB41800/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=1997
Oct  7 21:42:21.301: //-1/80DCADB41800/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
Oct  7 21:42:21.301: //-1/80DCADB41800/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
 1: Dial-peer Tag=101
Oct  7 21:42:21.345: //-1//DPM/dpMatchPeersCore:
   Calling Number=, Called Number=8884343, Peer Info Type=DIALPEER_INFO_SPEECH
Oct  7 21:42:21.345: //-1//DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=8884343
Oct  7 21:42:21.345: //-1//DPM/dpMatchPeersCore:
   No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)
Oct  7 21:42:21.345: //-1//DPM/dpMatchPeers:
   Result=NO_MATCH(-1)
Oct  7 21:42:21.345: //-1//DPM/dpMatchPeersCore:
   Calling Number=, Called Number=1997, Peer Info Type=DIALPEER_INFO_SPEECH
Oct  7 21:42:21.345: //-1//DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=1997
Oct  7 21:42:21.345: //-1//DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
Oct  7 21:42:21.345: //-1//DPM/dpMatchPeers:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
 1: Dial-peer Tag=101

 
 
You see there he is trying to call 1997 which matches dial-peer 101 and sends 
him to the callmanager. this is expected behaviour as 1997 is the MVA number.
 
But he is also trying to make a ca

Re: [OSL | CCIE_Voice] HW conference with CAC problem

2010-10-07 Thread Prashant Patel

Hi,

Do you have any shared lines in your scenario across the two phones you are
conferencing ie shared lines between HQ , A or B phones? If so every call to
shared line needs 24kbps across the WAN.

HTH
Prashant

On Thu, Oct 7, 2010 at 2:16 PM, Roig Borrell, Francesc Xavier <
francesc.ro...@tecnocom.es> wrote:

>  Hi all,
>
>
>
> I have this scenario
>
>
>
> Hardware CFB configured  in HQ
>
> Ad-hoc conferene started by hq phone1 with br1 phone1 and br1 phone2. No
> problem with this
>
>
>
> HQ#SH sccp connections
>
> sess_idconn_idstype mode codec   ripaddr rport sport
>
> 33556436   33554594   conf  sendrecv g729b   192.168.21.327628 17266
>
> 33556436   33554590   conf  sendrecv g711u   192.168.20.427408 16608
>
> 33556436   33554588   conf  sendrecv g729b   192.168.20.227820 18172
>
> Total number of active session(s) 1, and connection(s) 3
>
>
>
> The problems starts if I configure CAC in BR1
>
> With br1 Location=48 , the  conference fails  (Message in hq phone1 Cannot
> complete the conference)
>
> I have to configure at least Location=72 in order to make it work.
>
> I have two g729 calls so with 48 should work. Any ideas why I need to
> configure 72?
>
>
>
> Thanks in advance!!
>
> Xavi
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

[OSL | CCIE_Voice] HW conference with CAC problem

2010-10-07 Thread Roig Borrell, Francesc Xavier

Hi all,

I have this scenario

Hardware CFB configured  in HQ
Ad-hoc conferene started by hq phone1 with br1 phone1 and br1 phone2. No 
problem with this

HQ#SH sccp connections
sess_idconn_idstype mode codec   ripaddr rport sport
33556436   33554594   conf  sendrecv g729b   192.168.21.327628 17266
33556436   33554590   conf  sendrecv g711u   192.168.20.427408 16608
33556436   33554588   conf  sendrecv g729b   192.168.20.227820 18172
Total number of active session(s) 1, and connection(s) 3

The problems starts if I configure CAC in BR1
With br1 Location=48 , the  conference fails  (Message in hq phone1 Cannot 
complete the conference)
I have to configure at least Location=72 in order to make it work.
I have two g729 calls so with 48 should work. Any ideas why I need to configure 
72?

Thanks in advance!!
Xavi

___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] Help

2010-10-07 Thread Iwan Hoogendoorn

Hi,

My procedure is usually to just enable autoregistration for only the subscriber 
...
Let your phones register ...
You will see the phones that are of IPexpert and your own phones.
Once your own phones are registered you can disable autoregistration and delete 
the IPexpert phones and leave your own phones in.
If your phones does not autoregister ... then you have another problem.

Thanks,
Iwan Hoogendoorn

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ömer ketene
Sent: donderdag 7 oktober 2010 16:53
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Help

Hello ,

Yesterday While I was practicing voice Workbook 1 Lab 4A I saw that phones 
belonging to the HQ PHONE2 and BR1 PHone2 are preconfigured on call manager as 
device types of 7960 and CIPC.I have two 7965 for HQPHONE2 and BR1Phone2 in my 
site.Is there any easy way to transfer the configuration of 7960 and CIPC to my 
 7965's ?.

I tried changing mac addresses of preconfigured 7960 and CIPC but as device 
types doesnt match with 7965's at my site it was not a solution to this problem.

Finally,as there was not much configuration on call manager I just added my 
7965's to the system and made the same configuration that you made for 7960 and 
CIPC step by step on them, However in future complex labs this may cause such a 
huge loss of time to transfer all preconfigured configuration to my 7965's by 
this way .

What could be the solution for this?

Thanks,
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Re: [OSL | CCIE_Voice] CCIE Remote LAB

2010-10-07 Thread Iwan Hoogendoorn

Hi,

When you book the remote racks on proctorlabs.com there is no limitation at all.
The only thing you need is a few hardphones and a router and a switch to set up 
the EzVPN connection in order to register your own phones.

If that does not answer your question please let me know!
Iwan Hoogendoorn


From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Saeed IDris
Sent: woensdag 6 oktober 2010 22:44
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] CCIE Remote LAB

Hi everyone,
I'm seeking for a advice, I'm planning to prepare for the lab, if I book remote 
CCIE lab, what is kind of limitation and what is work around (product)?
I'm planning to use in my lab:

-  Cisco ISR Routers *4 (Router HQ, Router BR1, Router BR2 & Router 
PSTN).

-  Servers (CUCM, Unity Connection, Presence, UCCX).

-  Interface Telephony (E1 , T1)

-  Cisco IP Communicator (I will run it from Virtual machine).

Regards,

SAEED
___
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Re: [OSL | CCIE_Voice] UCCX challenges

2010-10-07 Thread George Goglidze

Hi,

You might be using some feature that is not supported by your license...
What license do you have and what features are you using???

Cheers,

On Wed, Oct 6, 2010 at 11:25 PM, CCIE Voice GMAIL <
givemeccievoice2...@gmail.com> wrote:

>  I know that sometimes the thing that fixes problems like this for me is
> going to the Control Center and restarting the UCCX Engine.  It’s obviously
> not ideal in the actual lab, but if it fixes it, run with it.
>
>
>
> Also, as someone before me suggested, you should really do a Reactive
> Debug.  This will point out your problem most of the time.
>
>
>
> Hope this helps,
>
> Jeff
>
>
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *ayman labib
> *Sent:* Wednesday, October 06, 2010 2:31 PM
>
> *To:* ccie_voice@onlinestudylist.com
> *Cc:* pithog...@yahoo.com
>
> *Subject:* Re: [OSL | CCIE_Voice] UCCX challenges
>
>
>
> Check your CSQ field if it's spelled properly and have quotations. That's
> what got me the other time.
>
>
>
>
>  --
>
> *From:* "cciefo...@hotmail.com" 
> *To:* Pithog Oil ;
> ccie_voice-boun...@onlinestudylist.com; ccie_voice@onlinestudylist.com
> *Sent:* Wed, October 6, 2010 5:27:27 PM
> *Subject:* Re: [OSL | CCIE_Voice] UCCX challenges
>
> There is something wrong with the script.  It could be a misspelled
> variable; or if you are referencing a holiday script that coukd be wrong
> too.  Try doing. Reactive debug to see where the problem is in the script.
> Sent from my Verizon Wireless BlackBerry
>
> -Original Message-
> From: Pithog Oil 
> Sender: ccie_voice-boun...@onlinestudylist.com
> Date: Wed, 6 Oct 2010 14:12:27
> To: 
> Subject: [OSL | CCIE_Voice] UCCX challenges
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
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Re: [OSL | CCIE_Voice] Help

2010-10-07 Thread Steve Denney (stdenney)

Hi Omer,

 

Easiest approach to deal with this is to do what you did - insert your 7965s 
manually into the CUCM database. But then do a BAT export of those two phones, 
and save the file locally on your workstation. Then you'll be able to quickly 
upload the phones at the start of each lab session.

 

cheers, sd

 

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ömer ketene
Sent: Thursday, October 07, 2010 10:53 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Help

 

Hello ,

Yesterday While I was practicing voice Workbook 1 Lab 4A I saw that phones 
belonging to the HQ PHONE2 and BR1 PHone2 are preconfigured on call manager as 
device types of 7960 and CIPC.I have two 7965 for HQPHONE2 and BR1Phone2 in my 
site.Is there any easy way to transfer the configuration of 7960 and CIPC to my 
 7965's ?.

I tried changing mac addresses of preconfigured 7960 and CIPC but as device 
types doesnt match with 7965's at my site it was not a solution to this problem.

Finally,as there was not much configuration on call manager I just added my 
7965's to the system and made the same configuration that you made for 7960 and 
CIPC step by step on them, However in future complex labs this may cause such a 
huge loss of time to transfer all preconfigured configuration to my 7965's by 
this way .

What could be the solution for this?

Thanks,

___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

[OSL | CCIE_Voice] Help

2010-10-07 Thread ömer ketene


Hello ,



Yesterday While I was practicing voice Workbook 1 Lab 4A I saw that phones
belonging to the HQ PHONE2 and BR1 PHone2 are preconfigured on call manager as
device types of 7960 and CIPC.I have two 7965 for HQPHONE2 and BR1Phone2 in my
site.Is there any easy way to transfer the configuration of 7960 and CIPC to
my  7965's ?.



I tried changing mac addresses of preconfigured 7960 and CIPC but as device
types doesnt match with 7965's at my site it was not a solution to this
problem.



Finally,as there was not much configuration on call manager I just added my
7965's to the system and made the same configuration that you made for 7960 and
CIPC step by step on them, However in future complex labs this may cause such a
huge loss of time to transfer all preconfigured configuration to my 7965's by
this way .



What could be the solution for this?

Thanks,

  ___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

[OSL | CCIE_Voice] RE : Volume 2 Lab 1 4.1 Call issue - xcoder

2010-10-07 Thread Friderich Claude

Another thing
 
Incoming call leg to CME is G711 (DP incoming) as resquested
if default codec for the sip phone is registered on cme (default G.729), you 
will need a xcoder on cme
Or just put a codec g711ulaw on the voice register pool and it gonna work 
withtout any xcoder on the cme
 
Regards

 


De: ccie_voice-boun...@onlinestudylist.com de la part de Friderich Claude
Date: jeu. 10/7/2010 3:15
À: Amr Sherif; ccie_voice@onlinestudylist.com
Objet : Re: [OSL | CCIE_Voice] Volume 2 Lab 1 4.1 Call issue - xcoder



Hi,

 

Did you put a xcoder on hq-rtr ?

 

br2ph (CME) < hq-rtr(GK) <-- CCM

zone UCME G711CUBEG729 zone UCM

 

As you can see you're involving a different codec for the inbound call leg and 
outbound call leg at the hq-rtr

 

And CUBE doesn't support handshake TCS at all

So you have to disable 'Wait for H245 capacity set on cucm trunk as well

 

Hope this gonna help ...

 

Regards

 

 

Claude Friderich

PreSales Support

 
ccvp_voice_sm
 

NETCORE PSF S.A.

49 rue du Baerendall

B.P.65 L-8201 Mamer

Téléphone: 31 33 80-407

Fax: 31 33 80 8-407

GSM: 621 303 616

E-mail: cfrider...@netcore.lu  

 

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Amr Sherif
Sent: jeudi 7 octobre 2010 13:40
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Volume 2 Lab 1 4.1 Call issue

 


Hello,
 
I have any issue is that when making a call from HQ Phones to BR2 Phones SCCP 
or SIP,the call keeps ringing even after i pick up the call.
 
I check the same configuration as is done in Proctor Guide but still cannot 
make it work, any one face that issue?.



Best regards,






 
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This email was Anti Virus checked by Astaro Security Gateway. 
<>___
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Re: [OSL | CCIE_Voice] Volume 2 Lab 1 4.1 Call issue

2010-10-07 Thread Paul Kruger

Yeah. What Tam said. Just to note: the MRGL used for your trunk should
include the IOS MTP you configured, and this MUST be registered with CUCM.
Keep this in mind, since the Xcoder would be registered to CUCME on HQ-RTR.

On Thu, Oct 7, 2010 at 2:14 PM, Tam Nhu  wrote:

> Hi Amr,
>
> since you are doing lab 1, and it has CUBE involved right.  So make sure
> the following need to check and configure
>
> Configure Xcoder and IOS MTP on HQ
> On the trunk, make sure to
>
>- Checked MTP
>- Unchecked  'Wait for H245 capacity set'
>- Checked Enable Inbound Fast Start
>
> Hope that help,
> TN.
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
___
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Re: [OSL | CCIE_Voice] Volume 2 Lab 1 4.1 Call issue - xcoder

2010-10-07 Thread Friderich Claude

Hi,

 

Did you put a xcoder on hq-rtr ?

 

br2ph (CME) < hq-rtr(GK) <-- CCM

zone UCME G711CUBEG729 zone UCM

 

As you can see you’re involving a different codec for the inbound call leg and 
outbound call leg at the hq-rtr

 

And CUBE doesn’t support handshake TCS at all

So you have to disable 'Wait for H245 capacity set on cucm trunk as well

 

Hope this gonna help …

 

Regards

 

 

Claude Friderich

PreSales Support

 

NETCORE PSF S.A.

49 rue du Baerendall

B.P.65 L-8201 Mamer

Téléphone: 31 33 80-407

Fax: 31 33 80 8-407

GSM: 621 303 616

E-mail: cfrider...@netcore.lu  

 

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Amr Sherif
Sent: jeudi 7 octobre 2010 13:40
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Volume 2 Lab 1 4.1 Call issue

 

 
Hello,
 
I have any issue is that when making a call from HQ Phones to BR2 Phones SCCP 
or SIP,the call keeps ringing even after i pick up the call.
 
I check the same configuration as is done in Proctor Guide but still cannot 
make it work, any one face that issue?.



Best regards,






 
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This email was Anti Virus checked by Astaro Security Gateway. 
<>___
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Re: [OSL | CCIE_Voice] Volume 2 Lab 1 4.1 Call issue

2010-10-07 Thread Friderich Claude

Hi,

 

Did you put a codec on hq-rtr ?

 

br2ph (CME) < hq-rtr(GK) <-- CCM

zone UCME G711CUBEG729 zone UCM

 

As you can see you’re involving a different codec for the inbound call leg and 
outbound call leg at the hq-rtr

 

And CUBE doesn’t support handshake TCS at all

So you have to disable 'Wait for H245 capacity set on cucm trunk as well

 

Hope this gonna help …

 

Regards

 

 

Claude Friderich

PreSales Support

 

NETCORE PSF S.A.

49 rue du Baerendall

B.P.65 L-8201 Mamer

Téléphone: 31 33 80-407

Fax: 31 33 80 8-407

GSM: 621 303 616

E-mail: cfrider...@netcore.lu  

 

From: ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Amr Sherif
Sent: jeudi 7 octobre 2010 13:40
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Volume 2 Lab 1 4.1 Call issue

 

 
Hello,
 
I have any issue is that when making a call from HQ Phones to BR2 Phones SCCP 
or SIP,the call keeps ringing even after i pick up the call.
 
I check the same configuration as is done in Proctor Guide but still cannot 
make it work, any one face that issue?.



Best regards,






 
-- 
This email was Anti Virus checked by Astaro Security Gateway. 
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Re: [OSL | CCIE_Voice] Volume 2 Lab 1 4.1 Call issue

2010-10-07 Thread Tam Nhu

Hi Amr,

since you are doing lab 1, and it has CUBE involved right.  So make sure the
following need to check and configure

Configure Xcoder and IOS MTP on HQ
On the trunk, make sure to

   - Checked MTP
   - Unchecked  'Wait for H245 capacity set'
   - Checked Enable Inbound Fast Start

Hope that help,
TN.
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Re: [OSL | CCIE_Voice] MVA Troubleshooting lab 6 question 5.3

2010-10-07 Thread amr thabt

Hi ,
can you post your dialpeer configuration

hth
amr

On Thu, Oct 7, 2010 at 12:24 AM, Pithog Oil  wrote:

>
> I spent some time trying to figure out a fix but, i have not gotten the
> solution yet, whenever i call "*2123945010"* in an attemp to invoke my MVA
> application, it rings quite fine and prompts me for my remote destination,
> ID and when i press 1 to call an extension, i try to place a call but the
> call gets *dropped*.
> **
> I have my MVA number specified on UCM to be 5010, i will appreciate
> assistance on how to fix thas issue,
>
> i think  a translation profile was used in  the solutions to translate
> /5002/ /2123942123/
> but its not clear how the translation pattern was invoked.
>
>  Thanks in Anticipation.
>
>
>
>
> ___
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
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Re: [OSL | CCIE_Voice] Volume 2 Lab 1 4.1 Call issue

2010-10-07 Thread Stutz, Bernhard

Hi Amr,

 

Its being send via the Gatekeeper right?

Sounds like the usual codec issue...

Check your complete call flow if you have the the same codec and if
there is a transcoder needed.

 

If this doesn't help provide some more details please.

 

Hth,

Bernhard

 

Von: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] Im Auftrag von Amr
Sherif
Gesendet: Donnerstag, 7. Oktober 2010 13:40
An: ccie_voice@onlinestudylist.com
Betreff: [OSL | CCIE_Voice] Volume 2 Lab 1 4.1 Call issue

 

 
Hello,
 
I have any issue is that when making a call from HQ Phones to BR2 Phones
SCCP or SIP,the call keeps ringing even after i pick up the call.
 
I check the same configuration as is done in Proctor Guide but still
cannot make it work, any one face that issue?.



Best regards,






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[OSL | CCIE_Voice] Volume 2 Lab 1 4.1 Call issue

2010-10-07 Thread Amr Sherif


 
Hello,
 
I have any issue is that when making a call from HQ Phones to BR2 Phones SCCP 
or SIP,the call keeps ringing even after i pick up the call.
 
I check the same configuration as is done in Proctor Guide but still cannot 
make it work, any one face that issue?.



Best regards,




  ___
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[OSL | CCIE_Voice] Question on H323 Trunks & Media Negotiation

2010-10-07 Thread Mann Chaddha

Hi Everyone

I need some clarity on a call flow that uses multiple ICT/H323 Trunks.
Here is the scenario:

Cluster 1
Cluster 2
Cluster 3
   WAN
   WAN
PH 1 ---> GK Trunk (G729 Region) --> PH 2 (VM Roll Over)
---> ICT Trunk (G711 Region Only) > VM Ports
(VG248) to Octel

The call originates from Ph1 which uses a GK Trunk to arrive at
Cluster 2 Ph2. When the Ph2 call rolls over to VM, then the VM Pilot
points to a RP associated with an ICT with G711 Region only. My
understanding is that the ICT will still negotiate the calls G729 ( <
80 kbps) & forward it to Cluster 3 and it will not invoke any XCoder.
Cluster 3, I believe, will forward the call to VM Ports as G729 only.

I want to hard-code the VM Ports to talk only G711 but in this case,
even with appropriate Region Settings, I am unable to.

Is my understanding correct or am I missing something here?

Kindly suggest.

Thanks
Mann
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[OSL | CCIE_Voice] RE : Volume 2 Lab 1 4.1 Call issue - xcoder

Re: [OSL | CCIE_Voice] Volume 2 Lab 1 4.1 Call issue

Re: [OSL | CCIE_Voice] Volume 2 Lab 1 4.1 Call issue - xcoder

Re: [OSL | CCIE_Voice] Volume 2 Lab 1 4.1 Call issue

Re: [OSL | CCIE_Voice] Volume 2 Lab 1 4.1 Call issue

Re: [OSL | CCIE_Voice] MVA Troubleshooting lab 6 question 5.3

Re: [OSL | CCIE_Voice] Volume 2 Lab 1 4.1 Call issue

[OSL | CCIE_Voice] Volume 2 Lab 1 4.1 Call issue

[OSL | CCIE_Voice] Question on H323 Trunks & Media Negotiation

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